From a9803b9f75c07b97c597cca6223e29887a75b759 Mon Sep 17 00:00:00 2001 From: melanson Date: Sat, 30 Mar 2002 22:27:45 +0000 Subject: reworked ADPCM decoders; changes include: * fixed MS IMA ADPCM * dissolved adpcm.c/.h into appropriate ad_* decoders * DK4 audio is handled directly by IMA ADPCM decoder (this obsoletes ad_dk4adpcm.c) git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@5409 b3059339-0415-0410-9bf9-f77b7e298cf2 --- adpcm.c | 66 ++++++++- adpcm.h | 4 +- etc/codecs.conf | 2 +- libmpcodecs/Makefile | 2 +- libmpcodecs/ad.c | 2 - libmpcodecs/ad_dk3adpcm.c | 222 ++++++++++++++++++++++++++--- libmpcodecs/ad_imaadpcm.c | 351 +++++++++++++++++++++++++++++++++++++++++++--- libmpcodecs/ad_msadpcm.c | 148 ++++++++++++++++++- 8 files changed, 748 insertions(+), 49 deletions(-) diff --git a/adpcm.c b/adpcm.c index fc1e857653..c48afddc21 100644 --- a/adpcm.c +++ b/adpcm.c @@ -9,6 +9,7 @@ (C) 2001 Mike Melanson */ +#if 0 #include "config.h" #include "bswap.h" #include "adpcm.h" @@ -119,7 +120,7 @@ void decode_nibbles(unsigned short *output, } } -int ima_adpcm_decode_block(unsigned short *output, unsigned char *input, +int qt_ima_adpcm_decode_block(unsigned short *output, unsigned char *input, int channels) { int initial_predictor_l = 0; @@ -180,6 +181,67 @@ int ima_adpcm_decode_block(unsigned short *output, unsigned char *input, return IMA_ADPCM_SAMPLES_PER_BLOCK * channels; } +int ms_ima_adpcm_decode_block(unsigned short *output, unsigned char *input, + int channels, int block_size) +{ + int initial_predictor_l = 0; + int initial_predictor_r = 0; + int initial_index_l = 0; + int initial_index_r = 0; + int i; + + initial_predictor_l = BE_16(&input[0]); + initial_index_l = initial_predictor_l; + + // mask, sign-extend, and clamp the predictor portion + initial_predictor_l &= 0xFF80; + SE_16BIT(initial_predictor_l); + CLAMP_S16(initial_predictor_l); + + // mask and clamp the index portion + initial_index_l &= 0x7F; + CLAMP_0_TO_88(initial_index_l); + + // handle stereo + if (channels > 1) + { + initial_predictor_r = BE_16(&input[IMA_ADPCM_BLOCK_SIZE]); + initial_index_r = initial_predictor_r; + + // mask, sign-extend, and clamp the predictor portion + initial_predictor_r &= 0xFF80; + SE_16BIT(initial_predictor_r); + CLAMP_S16(initial_predictor_r); + + // mask and clamp the index portion + initial_index_r &= 0x7F; + CLAMP_0_TO_88(initial_index_r); + } + + // break apart all of the nibbles in the block + if (channels == 1) + for (i = 0; i < IMA_ADPCM_SAMPLES_PER_BLOCK / 2; i++) + { + output[i * 2 + 0] = input[2 + i] & 0x0F; + output[i * 2 + 1] = input[2 + i] >> 4; + } + else + for (i = 0; i < IMA_ADPCM_SAMPLES_PER_BLOCK / 2 * 2; i++) + { + output[i * 4 + 0] = input[2 + i] & 0x0F; + output[i * 4 + 1] = input[2 + IMA_ADPCM_BLOCK_SIZE + i] & 0x0F; + output[i * 4 + 2] = input[2 + i] >> 4; + output[i * 4 + 3] = input[2 + IMA_ADPCM_BLOCK_SIZE + i] >> 4; + } + + decode_nibbles(output, + IMA_ADPCM_SAMPLES_PER_BLOCK * channels, channels, + initial_predictor_l, initial_index_l, + initial_predictor_r, initial_index_r); + + return IMA_ADPCM_SAMPLES_PER_BLOCK * channels; +} + int ms_adpcm_decode_block(unsigned short *output, unsigned char *input, int channels, int block_size) { @@ -439,3 +501,5 @@ int dk3_adpcm_decode_block(unsigned short *output, unsigned char *input) return out_ptr; } +#endif + diff --git a/adpcm.h b/adpcm.h index 9fe7c8676a..e4048c4ca1 100644 --- a/adpcm.h +++ b/adpcm.h @@ -20,8 +20,10 @@ // this isn't exact #define DK3_ADPCM_SAMPLES_PER_BLOCK 6000 -int ima_adpcm_decode_block(unsigned short *output, unsigned char *input, +int qt_ima_adpcm_decode_block(unsigned short *output, unsigned char *input, int channels); +int ms_ima_adpcm_decode_block(unsigned short *output, unsigned char *input, + int channels, int block_size); int ms_adpcm_decode_block(unsigned short *output, unsigned char *input, int channels, int block_size); int dk4_adpcm_decode_block(unsigned short *output, unsigned char *input, diff --git a/etc/codecs.conf b/etc/codecs.conf index 0d08ed9d34..ec3e320959 100644 --- a/etc/codecs.conf +++ b/etc/codecs.conf @@ -904,7 +904,7 @@ audiocodec dk4adpcm status working comment "This format number was used by Duck Corp. but not officially registered with Microsoft" format 0x61 - driver dk4adpcm + driver imaadpcm audiocodec dk3adpcm info "Duck DK3 ADPCM (rogue format number)" diff --git a/libmpcodecs/Makefile b/libmpcodecs/Makefile index 5ac92114f9..c774513e93 100644 --- a/libmpcodecs/Makefile +++ b/libmpcodecs/Makefile @@ -3,7 +3,7 @@ include ../config.mak LIBNAME = libmpcodecs.a -AUDIO_SRCS=dec_audio.c ad.c ad_a52.c ad_acm.c ad_alaw.c ad_dk3adpcm.c ad_dk4adpcm.c ad_dshow.c ad_dvdpcm.c ad_ffmpeg.c ad_hwac3.c ad_imaadpcm.c ad_mp3.c ad_msadpcm.c ad_pcm.c ad_roqaudio.c ad_msgsm.c ad_faad.c +AUDIO_SRCS=dec_audio.c ad.c ad_a52.c ad_acm.c ad_alaw.c ad_dk3adpcm.c ad_dshow.c ad_dvdpcm.c ad_ffmpeg.c ad_hwac3.c ad_imaadpcm.c ad_mp3.c ad_msadpcm.c ad_pcm.c ad_roqaudio.c ad_msgsm.c ad_faad.c VIDEO_SRCS=dec_video.c vd.c vd_null.c vd_cinepak.c vd_qtrpza.c vd_ffmpeg.c vd_dshow.c vd_vfw.c vd_odivx.c vd_divx4.c vd_raw.c vd_xanim.c vd_msvidc.c vd_fli.c vd_qtrle.c vd_qtsmc.c vd_roqvideo.c vd_cyuv.c vd_nuv.c vd_libmpeg2.c vd_msrle.c vd_huffyuv.c vd_zlib.c ifeq ($(PNG),yes) diff --git a/libmpcodecs/ad.c b/libmpcodecs/ad.c index 2171405550..4ec4c1b15a 100644 --- a/libmpcodecs/ad.c +++ b/libmpcodecs/ad.c @@ -25,7 +25,6 @@ extern ad_functions_t mpcodecs_ad_dvdpcm; extern ad_functions_t mpcodecs_ad_alaw; extern ad_functions_t mpcodecs_ad_imaadpcm; extern ad_functions_t mpcodecs_ad_msadpcm; -extern ad_functions_t mpcodecs_ad_dk4adpcm; extern ad_functions_t mpcodecs_ad_dk3adpcm; extern ad_functions_t mpcodecs_ad_roqaudio; extern ad_functions_t mpcodecs_ad_dshow; @@ -47,7 +46,6 @@ ad_functions_t* mpcodecs_ad_drivers[] = &mpcodecs_ad_alaw, &mpcodecs_ad_imaadpcm, &mpcodecs_ad_msadpcm, - &mpcodecs_ad_dk4adpcm, &mpcodecs_ad_dk3adpcm, &mpcodecs_ad_roqaudio, &mpcodecs_ad_msgsm, diff --git a/libmpcodecs/ad_dk3adpcm.c b/libmpcodecs/ad_dk3adpcm.c index a6de3d96e1..d940ec3748 100644 --- a/libmpcodecs/ad_dk3adpcm.c +++ b/libmpcodecs/ad_dk3adpcm.c @@ -1,8 +1,19 @@ +/* + DK3 ADPCM Decoder for MPlayer + by Mike Melanson + + This file is responsible for decoding audio data encoded with + Duck Corp's DK3 ADPCM algorithm. Details about the data format + can be found here: + http://www.pcisys.net/~melanson/codecs/ +*/ + #include #include #include #include "config.h" +#include "bswap.h" #include "ad_internal.h" static ad_info_t info = @@ -17,27 +28,70 @@ static ad_info_t info = LIBAD_EXTERN(dk3adpcm) -#include "adpcm.h" +#define DK3_ADPCM_PREAMBLE_SIZE 16 -static int init(sh_audio_t *sh_audio) +#define LE_16(x) (le2me_16(*(unsigned short *)(x))) +#define LE_32(x) (le2me_32(*(unsigned int *)(x))) + +// useful macros +// clamp a number between 0 and 88 +#define CLAMP_0_TO_88(x) if (x < 0) x = 0; else if (x > 88) x = 88; +// clamp a number within a signed 16-bit range +#define CLAMP_S16(x) if (x < -32768) x = -32768; \ + else if (x > 32767) x = 32767; +// clamp a number above 16 +#define CLAMP_ABOVE_16(x) if (x < 16) x = 16; +// sign extend a 16-bit value +#define SE_16BIT(x) if (x & 0x8000) x -= 0x10000; +// sign extend a 4-bit value +#define SE_4BIT(x) if (x & 0x8) x -= 0x10; + +// pertinent tables +static int adpcm_step[89] = +{ + 7, 8, 9, 10, 11, 12, 13, 14, 16, 17, + 19, 21, 23, 25, 28, 31, 34, 37, 41, 45, + 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, + 130, 143, 157, 173, 190, 209, 230, 253, 279, 307, + 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, + 876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066, + 2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358, + 5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899, + 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767 +}; + +static int adpcm_index[16] = +{ + -1, -1, -1, -1, 2, 4, 6, 8, + -1, -1, -1, -1, 2, 4, 6, 8 +}; + +static int preinit(sh_audio_t *sh_audio) { - sh_audio->channels=sh_audio->wf->nChannels; - sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; - sh_audio->i_bps=DK3_ADPCM_BLOCK_SIZE* - (sh_audio->channels*sh_audio->samplerate) / DK3_ADPCM_SAMPLES_PER_BLOCK; + sh_audio->audio_out_minsize = sh_audio->wf->nBlockAlign * 6; + sh_audio->ds->ss_div = + (sh_audio->wf->nBlockAlign - DK3_ADPCM_PREAMBLE_SIZE) * 8 / 3; + sh_audio->ds->ss_mul = sh_audio->wf->nBlockAlign; return 1; } -static int preinit(sh_audio_t *sh_audio) +static int init(sh_audio_t *sh_audio) { - sh_audio->audio_out_minsize=DK3_ADPCM_SAMPLES_PER_BLOCK * 4; - sh_audio->ds->ss_div=DK3_ADPCM_SAMPLES_PER_BLOCK; - sh_audio->ds->ss_mul=DK3_ADPCM_BLOCK_SIZE; + sh_audio->channels = sh_audio->wf->nChannels; + sh_audio->samplerate = sh_audio->wf->nSamplesPerSec; + sh_audio->i_bps = + (sh_audio->ds->ss_mul * sh_audio->samplerate) / sh_audio->ds->ss_div; + + if ((sh_audio->a_in_buffer = + (unsigned char *)malloc(sh_audio->ds->ss_mul)) == NULL) + return 0; + return 1; } -static void uninit(sh_audio_t *sh) +static void uninit(sh_audio_t *sh_audio) { + free(sh_audio->a_in_buffer); } static int control(sh_audio_t *sh,int cmd,void* arg, ...) @@ -46,15 +100,143 @@ static int control(sh_audio_t *sh,int cmd,void* arg, ...) return CONTROL_UNKNOWN; } +#define DK3_GET_NEXT_NIBBLE() \ + if (decode_top_nibble_next) \ + { \ + nibble = (last_byte >> 4) & 0x0F; \ + decode_top_nibble_next = 0; \ + } \ + else \ + { \ + last_byte = input[in_ptr++]; \ + nibble = last_byte & 0x0F; \ + decode_top_nibble_next = 1; \ + } + +// note: This decoder assumes the format 0x62 data always comes in +// stereo flavor +static int dk3_adpcm_decode_block(unsigned short *output, unsigned char *input, + int block_size) +{ + int sum_pred; + int diff_pred; + int sum_index; + int diff_index; + int diff_channel; + int in_ptr = 0x10; + int out_ptr = 0; + + unsigned char last_byte = 0; + unsigned char nibble; + int decode_top_nibble_next = 0; + + // ADPCM work variables + int sign; + int delta; + int step; + int diff; + + sum_pred = LE_16(&input[10]); + diff_pred = LE_16(&input[12]); + SE_16BIT(sum_pred); + SE_16BIT(diff_pred); + diff_channel = diff_pred; + sum_index = input[14]; + diff_index = input[15]; + + while (in_ptr < block_size) +// while (in_ptr < 2048) + { + // process the first predictor of the sum channel + DK3_GET_NEXT_NIBBLE(); + + step = adpcm_step[sum_index]; + + sign = nibble & 8; + delta = nibble & 7; + + diff = step >> 3; + if (delta & 4) diff += step; + if (delta & 2) diff += step >> 1; + if (delta & 1) diff += step >> 2; + + if (sign) + sum_pred -= diff; + else + sum_pred += diff; + + CLAMP_S16(sum_pred); + + sum_index += adpcm_index[nibble]; + CLAMP_0_TO_88(sum_index); + + // process the diff channel predictor + DK3_GET_NEXT_NIBBLE(); + + step = adpcm_step[diff_index]; + + sign = nibble & 8; + delta = nibble & 7; + + diff = step >> 3; + if (delta & 4) diff += step; + if (delta & 2) diff += step >> 1; + if (delta & 1) diff += step >> 2; + + if (sign) + diff_pred -= diff; + else + diff_pred += diff; + + CLAMP_S16(diff_pred); + + diff_index += adpcm_index[nibble]; + CLAMP_0_TO_88(diff_index); + + // output the first pair of stereo PCM samples + diff_channel = (diff_channel + diff_pred) / 2; + output[out_ptr++] = sum_pred + diff_channel; + output[out_ptr++] = sum_pred - diff_channel; + + // process the second predictor of the sum channel + DK3_GET_NEXT_NIBBLE(); + + step = adpcm_step[sum_index]; + + sign = nibble & 8; + delta = nibble & 7; + + diff = step >> 3; + if (delta & 4) diff += step; + if (delta & 2) diff += step >> 1; + if (delta & 1) diff += step >> 2; + + if (sign) + sum_pred -= diff; + else + sum_pred += diff; + + CLAMP_S16(sum_pred); + + sum_index += adpcm_index[nibble]; + CLAMP_0_TO_88(sum_index); + + // output the second pair of stereo PCM samples + output[out_ptr++] = sum_pred + diff_channel; + output[out_ptr++] = sum_pred - diff_channel; + } + + return out_ptr; +} + static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen) { - int len=-1; - unsigned char ibuf[DK3_ADPCM_BLOCK_SIZE * 2]; /* bytes / stereo frame */ - if (demux_read_data(sh_audio->ds, ibuf, - DK3_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels) != - DK3_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels) - return len; /* EOF */ - len = 2 * dk3_adpcm_decode_block( - (unsigned short*)buf,ibuf); - return len; + if (demux_read_data(sh_audio->ds, sh_audio->a_in_buffer, + sh_audio->ds->ss_mul) != + sh_audio->ds->ss_mul) + return -1; /* EOF */ + + return 2 * dk3_adpcm_decode_block( + (unsigned short*)buf, sh_audio->a_in_buffer, + sh_audio->wf->nBlockAlign); } diff --git a/libmpcodecs/ad_imaadpcm.c b/libmpcodecs/ad_imaadpcm.c index eca064d6c5..7f9e076eb0 100644 --- a/libmpcodecs/ad_imaadpcm.c +++ b/libmpcodecs/ad_imaadpcm.c @@ -1,11 +1,71 @@ +/* + IMA ADPCM Decoder for MPlayer + by Mike Melanson + + This file is in charge of decoding all of the various IMA ADPCM data + formats that various entities have created. Details about the data + formats can be found here: + http://www.pcisys.net/~melanson/codecs/ + + So far, this file handles these formats: + 'ima4': IMA ADPCM found in QT files + 0x11: IMA ADPCM found in MS AVI/ASF/WAV files + 0x61: DK4 ADPCM found in certain AVI files on Sega Saturn CD-ROMs; + note that this is a 'rogue' format number in that it was + never officially registered with Microsoft +*/ + #include #include #include #include "config.h" +#include "bswap.h" #include "ad_internal.h" -#include "../adpcm.h" +#define MS_IMA_ADPCM_PREAMBLE_SIZE 4 + +#define QT_IMA_ADPCM_PREAMBLE_SIZE 2 +#define QT_IMA_ADPCM_BLOCK_SIZE 0x22 +#define QT_IMA_ADPCM_SAMPLES_PER_BLOCK 64 + +#define BE_16(x) (be2me_16(*(unsigned short *)(x))) +#define BE_32(x) (be2me_32(*(unsigned int *)(x))) +#define LE_16(x) (le2me_16(*(unsigned short *)(x))) +#define LE_32(x) (le2me_32(*(unsigned int *)(x))) + +// pertinent tables for IMA ADPCM +static int adpcm_step[89] = +{ + 7, 8, 9, 10, 11, 12, 13, 14, 16, 17, + 19, 21, 23, 25, 28, 31, 34, 37, 41, 45, + 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, + 130, 143, 157, 173, 190, 209, 230, 253, 279, 307, + 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, + 876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066, + 2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358, + 5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899, + 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767 +}; + +static int adpcm_index[16] = +{ + -1, -1, -1, -1, 2, 4, 6, 8, + -1, -1, -1, -1, 2, 4, 6, 8 +}; + +// useful macros +// clamp a number between 0 and 88 +#define CLAMP_0_TO_88(x) if (x < 0) x = 0; else if (x > 88) x = 88; +// clamp a number within a signed 16-bit range +#define CLAMP_S16(x) if (x < -32768) x = -32768; \ + else if (x > 32767) x = 32767; +// clamp a number above 16 +#define CLAMP_ABOVE_16(x) if (x < 16) x = 16; +// sign extend a 16-bit value +#define SE_16BIT(x) if (x & 0x8000) x -= 0x10000; +// sign extend a 4-bit value +#define SE_4BIT(x) if (x & 0x8) x -= 0x10; static ad_info_t info = { @@ -14,32 +74,52 @@ static ad_info_t info = AFM_IMAADPCM, "Nick Kurshev", "Mike Melanson", - "ima4 (MOV files)" + "" }; LIBAD_EXTERN(imaadpcm) +static int preinit(sh_audio_t *sh_audio) +{ + // not exactly sure what this field is for + sh_audio->audio_out_minsize = 8192; + + // if format is "ima4", assume the audio is coming from a QT file which + // indicates constant block size, whereas an AVI/ASF/WAV file will fill + // in this field with 0x11 + if ((sh_audio->format == 0x11) || (sh_audio->format == 0x61)) + { + sh_audio->ds->ss_div = (sh_audio->wf->nBlockAlign - + (MS_IMA_ADPCM_PREAMBLE_SIZE * sh_audio->wf->nChannels)) * 2; + sh_audio->ds->ss_mul = sh_audio->wf->nBlockAlign; + } + else + { + sh_audio->ds->ss_div = QT_IMA_ADPCM_SAMPLES_PER_BLOCK; + sh_audio->ds->ss_mul = QT_IMA_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels; + } + return 1; +} + static int init(sh_audio_t *sh_audio) { /* IMA-ADPCM 4:1 audio codec:*/ sh_audio->channels=sh_audio->wf->nChannels; sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; /* decodes 34 byte -> 64 short*/ - sh_audio->i_bps=IMA_ADPCM_BLOCK_SIZE*(sh_audio->channels*sh_audio->samplerate)/ - IMA_ADPCM_SAMPLES_PER_BLOCK; /* 1:4 */ - return 1; -} + sh_audio->i_bps = + (sh_audio->ds->ss_mul * sh_audio->samplerate) / sh_audio->ds->ss_div; + + if ((sh_audio->a_in_buffer = + (unsigned char *)malloc(sh_audio->ds->ss_mul)) == NULL) + return 0; -static int preinit(sh_audio_t *sh_audio) -{ - sh_audio->audio_out_minsize=4096; - sh_audio->ds->ss_div=IMA_ADPCM_SAMPLES_PER_BLOCK; - sh_audio->ds->ss_mul=IMA_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels; return 1; } -static void uninit(sh_audio_t *sh) +static void uninit(sh_audio_t *sh_audio) { + free(sh_audio->a_in_buffer); } static int control(sh_audio_t *sh,int cmd,void* arg, ...) @@ -48,12 +128,247 @@ static int control(sh_audio_t *sh,int cmd,void* arg, ...) return CONTROL_UNKNOWN; } +static void decode_nibbles(unsigned short *output, + int output_size, int channels, + int predictor_l, int index_l, + int predictor_r, int index_r) +{ + int step[2]; + int predictor[2]; + int index[2]; + int diff; + int i; + int sign; + int delta; + int channel_number = 0; + + step[0] = adpcm_step[index_l]; + step[1] = adpcm_step[index_r]; + predictor[0] = predictor_l; + predictor[1] = predictor_r; + index[0] = index_l; + index[1] = index_r; + + for (i = 0; i < output_size; i++) + { + delta = output[i]; + + index[channel_number] += adpcm_index[delta]; + CLAMP_0_TO_88(index[channel_number]); + + sign = delta & 8; + delta = delta & 7; + + diff = step[channel_number] >> 3; + if (delta & 4) diff += step[channel_number]; + if (delta & 2) diff += step[channel_number] >> 1; + if (delta & 1) diff += step[channel_number] >> 2; + + if (sign) + predictor[channel_number] -= diff; + else + predictor[channel_number] += diff; + + CLAMP_S16(predictor[channel_number]); + output[i] = predictor[channel_number]; + step[channel_number] = adpcm_step[index[channel_number]]; + + // toggle channel + channel_number ^= channels - 1; + + } +} + +static int qt_ima_adpcm_decode_block(unsigned short *output, + unsigned char *input, int channels) +{ + int initial_predictor_l = 0; + int initial_predictor_r = 0; + int initial_index_l = 0; + int initial_index_r = 0; + int i; + + initial_predictor_l = BE_16(&input[0]); + initial_index_l = initial_predictor_l; + + // mask, sign-extend, and clamp the predictor portion + initial_predictor_l &= 0xFF80; + SE_16BIT(initial_predictor_l); + CLAMP_S16(initial_predictor_l); + + // mask and clamp the index portion + initial_index_l &= 0x7F; + CLAMP_0_TO_88(initial_index_l); + + // handle stereo + if (channels > 1) + { + initial_predictor_r = BE_16(&input[QT_IMA_ADPCM_BLOCK_SIZE]); + initial_index_r = initial_predictor_r; + + // mask, sign-extend, and clamp the predictor portion + initial_predictor_r &= 0xFF80; + SE_16BIT(initial_predictor_r); + CLAMP_S16(initial_predictor_r); + + // mask and clamp the index portion + initial_index_r &= 0x7F; + CLAMP_0_TO_88(initial_index_r); + } + + // break apart all of the nibbles in the block + if (channels == 1) + for (i = 0; i < QT_IMA_ADPCM_SAMPLES_PER_BLOCK / 2; i++) + { + output[i * 2 + 0] = input[2 + i] & 0x0F; + output[i * 2 + 1] = input[2 + i] >> 4; + } + else + for (i = 0; i < QT_IMA_ADPCM_SAMPLES_PER_BLOCK / 2 * 2; i++) + { + output[i * 4 + 0] = input[2 + i] & 0x0F; + output[i * 4 + 1] = input[2 + QT_IMA_ADPCM_BLOCK_SIZE + i] & 0x0F; + output[i * 4 + 2] = input[2 + i] >> 4; + output[i * 4 + 3] = input[2 + QT_IMA_ADPCM_BLOCK_SIZE + i] >> 4; + } + + decode_nibbles(output, + QT_IMA_ADPCM_SAMPLES_PER_BLOCK * channels, channels, + initial_predictor_l, initial_index_l, + initial_predictor_r, initial_index_r); + + return QT_IMA_ADPCM_SAMPLES_PER_BLOCK * channels; +} + +static int ms_ima_adpcm_decode_block(unsigned short *output, + unsigned char *input, int channels, int block_size) +{ + int predictor_l = 0; + int predictor_r = 0; + int index_l = 0; + int index_r = 0; + int i; + int channel_counter; + int channel_index; + int channel_index_l; + int channel_index_r; + + predictor_l = LE_16(&input[0]); + SE_16BIT(predictor_l); + index_l = input[2]; + if (channels == 2) + { + predictor_r = LE_16(&input[4]); + SE_16BIT(predictor_r); + index_r = input[6]; + } + + if (channels == 1) + for (i = 0; + i < (block_size - MS_IMA_ADPCM_PREAMBLE_SIZE * channels) / 2; i++) + { + output[i * 2 + 0] = input[MS_IMA_ADPCM_PREAMBLE_SIZE + i] & 0x0F; + output[i * 2 + 1] = input[MS_IMA_ADPCM_PREAMBLE_SIZE + i] >> 4; + } + else + { + // encoded as 8 nibbles (4 bytes) per channel; switch channel every + // 4th byte + channel_counter = 0; + channel_index_l = 0; + channel_index_r = 1; + channel_index = channel_index_l; + for (i = 0; + i < (block_size - MS_IMA_ADPCM_PREAMBLE_SIZE * channels); i++) + { + output[channel_index + 0] = + input[MS_IMA_ADPCM_PREAMBLE_SIZE * 2 + i] & 0x0F; + output[channel_index + 2] = + input[MS_IMA_ADPCM_PREAMBLE_SIZE * 2 + i] >> 4; + channel_index += 4; + channel_counter++; + if (channel_counter == 4) + { + channel_index_l = channel_index; + channel_index = channel_index_r; + } + else if (channel_counter == 8) + { + channel_index_r = channel_index; + channel_index = channel_index_l; + channel_counter = 0; + } + } + } + + decode_nibbles(output, + (block_size - MS_IMA_ADPCM_PREAMBLE_SIZE * channels) * 2, + channels, + predictor_l, index_l, + predictor_r, index_r); + + return (block_size - MS_IMA_ADPCM_PREAMBLE_SIZE * channels) * 2; +} + +static int dk4_ima_adpcm_decode_block(unsigned short *output, + unsigned char *input, int channels, int block_size) +{ + int i; + int output_ptr; + int predictor_l = 0; + int predictor_r = 0; + int index_l = 0; + int index_r = 0; + + // the first predictor value goes straight to the output + predictor_l = output[0] = LE_16(&input[0]); + SE_16BIT(predictor_l); + index_l = input[2]; + if (channels == 2) + { + predictor_r = output[1] = LE_16(&input[4]); + SE_16BIT(predictor_r); + index_r = input[6]; + } + + output_ptr = channels; + for (i = MS_IMA_ADPCM_PREAMBLE_SIZE * channels; i < block_size; i++) + { + output[output_ptr++] = input[i] >> 4; + output[output_ptr++] = input[i] & 0x0F; + } + + decode_nibbles(&output[channels], + (block_size - MS_IMA_ADPCM_PREAMBLE_SIZE * channels) * 2 - channels, + channels, + predictor_l, index_l, + predictor_r, index_r); + + return (block_size - MS_IMA_ADPCM_PREAMBLE_SIZE * channels) * 2 - channels; +} + static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen) { - unsigned char ibuf[IMA_ADPCM_BLOCK_SIZE * 2]; /* bytes / stereo frame */ - if (demux_read_data(sh_audio->ds, ibuf, - IMA_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels) != - IMA_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels) - return -1; - return 2*ima_adpcm_decode_block((unsigned short*)buf,ibuf, sh_audio->wf->nChannels); + if (demux_read_data(sh_audio->ds, sh_audio->a_in_buffer, + sh_audio->ds->ss_mul) != + sh_audio->ds->ss_mul) + return -1; + + if (sh_audio->format == 0x11) + { + return 2 * ms_ima_adpcm_decode_block( + (unsigned short*)buf, sh_audio->a_in_buffer, sh_audio->wf->nChannels, + sh_audio->ds->ss_mul); + } + else if (sh_audio->format == 0x61) + { + return 2 * dk4_ima_adpcm_decode_block( + (unsigned short*)buf, sh_audio->a_in_buffer, sh_audio->wf->nChannels, + sh_audio->ds->ss_mul); + } + else + { + return 2 * qt_ima_adpcm_decode_block( + (unsigned short*)buf, sh_audio->a_in_buffer, sh_audio->wf->nChannels); + } } diff --git a/libmpcodecs/ad_msadpcm.c b/libmpcodecs/ad_msadpcm.c index 8d1b5cebcd..4a56c4c1fc 100644 --- a/libmpcodecs/ad_msadpcm.c +++ b/libmpcodecs/ad_msadpcm.c @@ -1,8 +1,18 @@ +/* + MS ADPCM Decoder for MPlayer + by Mike Melanson + + This file is responsible for decoding Microsoft ADPCM data. + Details about the data format can be found here: + http://www.pcisys.net/~melanson/codecs/ +*/ + #include #include #include #include "config.h" +#include "bswap.h" #include "ad_internal.h" static ad_info_t info = @@ -17,8 +27,40 @@ static ad_info_t info = LIBAD_EXTERN(msadpcm) +static int ms_adapt_table[] = +{ + 230, 230, 230, 230, 307, 409, 512, 614, + 768, 614, 512, 409, 307, 230, 230, 230 +}; + +static int ms_adapt_coeff1[] = +{ + 256, 512, 0, 192, 240, 460, 392 +}; + +static int ms_adapt_coeff2[] = +{ + 0, -256, 0, 64, 0, -208, -232 +}; + #define MS_ADPCM_PREAMBLE_SIZE 7 +#define LE_16(x) (le2me_16(*(unsigned short *)(x))) +#define LE_32(x) (le2me_32(*(unsigned int *)(x))) + +// useful macros +// clamp a number between 0 and 88 +#define CLAMP_0_TO_88(x) if (x < 0) x = 0; else if (x > 88) x = 88; +// clamp a number within a signed 16-bit range +#define CLAMP_S16(x) if (x < -32768) x = -32768; \ + else if (x > 32767) x = 32767; +// clamp a number above 16 +#define CLAMP_ABOVE_16(x) if (x < 16) x = 16; +// sign extend a 16-bit value +#define SE_16BIT(x) if (x & 0x8000) x -= 0x10000; +// sign extend a 4-bit value +#define SE_4BIT(x) if (x & 0x8) x -= 0x10; + static int preinit(sh_audio_t *sh_audio) { sh_audio->audio_out_minsize = sh_audio->wf->nBlockAlign * 4; @@ -54,14 +96,110 @@ static int control(sh_audio_t *sh,int cmd,void* arg, ...) return CONTROL_UNKNOWN; } +static int ms_adpcm_decode_block(unsigned short *output, unsigned char *input, + int channels, int block_size) +{ + int current_channel = 0; + int idelta[2]; + int sample1[2]; + int sample2[2]; + int coeff1[2]; + int coeff2[2]; + int stream_ptr = 0; + int out_ptr = 0; + int upper_nibble = 1; + int nibble; + int snibble; // signed nibble + int predictor; + + // fetch the header information, in stereo if both channels are present + if (input[stream_ptr] > 6) + mp_msg(MSGT_DECAUDIO, MSGL_WARN, + "MS ADPCM: coefficient (%d) out of range (should be [0..6])\n", + input[stream_ptr]); + coeff1[0] = ms_adapt_coeff1[input[stream_ptr]]; + coeff2[0] = ms_adapt_coeff2[input[stream_ptr]]; + stream_ptr++; + if (channels == 2) + { + if (input[stream_ptr] > 6) + mp_msg(MSGT_DECAUDIO, MSGL_WARN, + "MS ADPCM: coefficient (%d) out of range (should be [0..6])\n", + input[stream_ptr]); + coeff1[1] = ms_adapt_coeff1[input[stream_ptr]]; + coeff2[1] = ms_adapt_coeff2[input[stream_ptr]]; + stream_ptr++; + } + + idelta[0] = LE_16(&input[stream_ptr]); + stream_ptr += 2; + SE_16BIT(idelta[0]); + if (channels == 2) + { + idelta[1] = LE_16(&input[stream_ptr]); + stream_ptr += 2; + SE_16BIT(idelta[1]); + } + + sample1[0] = LE_16(&input[stream_ptr]); + stream_ptr += 2; + SE_16BIT(sample1[0]); + if (channels == 2) + { + sample1[1] = LE_16(&input[stream_ptr]); + stream_ptr += 2; + SE_16BIT(sample1[1]); + } + + sample2[0] = LE_16(&input[stream_ptr]); + stream_ptr += 2; + SE_16BIT(sample2[0]); + if (channels == 2) + { + sample2[1] = LE_16(&input[stream_ptr]); + stream_ptr += 2; + SE_16BIT(sample2[1]); + } + + while (stream_ptr < block_size) + { + // get the next nibble + if (upper_nibble) + nibble = snibble = input[stream_ptr] >> 4; + else + nibble = snibble = input[stream_ptr++] & 0x0F; + upper_nibble ^= 1; + SE_4BIT(snibble); + + predictor = ( + ((sample1[current_channel] * coeff1[current_channel]) + + (sample2[current_channel] * coeff2[current_channel])) / 256) + + (snibble * idelta[current_channel]); + CLAMP_S16(predictor); + sample2[current_channel] = sample1[current_channel]; + sample1[current_channel] = predictor; + output[out_ptr++] = predictor; + + // compute the next adaptive scale factor (a.k.a. the variable idelta) + idelta[current_channel] = + (ms_adapt_table[nibble] * idelta[current_channel]) / 256; + CLAMP_ABOVE_16(idelta[current_channel]); + + // toggle the channel + current_channel ^= channels - 1; + } + + return (block_size - (MS_ADPCM_PREAMBLE_SIZE * channels)) * 2; +} + static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen) { if (demux_read_data(sh_audio->ds, sh_audio->a_in_buffer, - sh_audio->ds->ss_mul) != - sh_audio->ds->ss_mul) - return -1; /* EOF */ + sh_audio->ds->ss_mul) != + sh_audio->ds->ss_mul) + return -1; /* EOF */ return 2 * ms_adpcm_decode_block( - (unsigned short*)buf, sh_audio->a_in_buffer, - sh_audio->wf->nChannels, sh_audio->wf->nBlockAlign); + (unsigned short*)buf, sh_audio->a_in_buffer, + sh_audio->wf->nChannels, sh_audio->wf->nBlockAlign); } -- cgit v1.2.3