Mercurial > libavcodec.hg
changeset 11077:c5f2e2abb70e libavcodec
Add ALAC 24 bps decoding support.
| author | jai_menon |
|---|---|
| date | Thu, 04 Feb 2010 16:21:26 +0000 |
| parents | 4a95670ed5f1 |
| children | e1eb8879e368 |
| files | alac.c |
| diffstat | 1 files changed, 103 insertions(+), 16 deletions(-) [+] |
line wrap: on
line diff
--- a/alac.c Thu Feb 04 15:30:37 2010 +0000 +++ b/alac.c Thu Feb 04 16:21:26 2010 +0000 @@ -77,6 +77,8 @@ int32_t *outputsamples_buffer[MAX_CHANNELS]; + int32_t *wasted_bits_buffer[MAX_CHANNELS]; + /* stuff from setinfo */ uint32_t setinfo_max_samples_per_frame; /* 0x1000 = 4096 */ /* max samples per frame? */ uint8_t setinfo_sample_size; /* 0x10 */ @@ -85,6 +87,7 @@ uint8_t setinfo_rice_kmodifier; /* 0x0e */ /* end setinfo stuff */ + int wasted_bits; } ALACContext; static void allocate_buffers(ALACContext *alac) @@ -96,6 +99,8 @@ alac->outputsamples_buffer[chan] = av_malloc(alac->setinfo_max_samples_per_frame * 4); + + alac->wasted_bits_buffer[chan] = av_malloc(alac->setinfo_max_samples_per_frame * 4); } } @@ -398,6 +403,56 @@ } } +static void decorrelate_stereo_24(int32_t *buffer[MAX_CHANNELS], + int32_t *buffer_out, + int32_t *wasted_bits_buffer[MAX_CHANNELS], + int wasted_bits, + int numchannels, int numsamples, + uint8_t interlacing_shift, + uint8_t interlacing_leftweight) +{ + int i; + + if (numsamples <= 0) + return; + + /* weighted interlacing */ + if (interlacing_leftweight) { + for (i = 0; i < numsamples; i++) { + int32_t a, b; + + a = buffer[0][i]; + b = buffer[1][i]; + + a -= (b * interlacing_leftweight) >> interlacing_shift; + b += a; + + if (wasted_bits) { + b = (b << wasted_bits) | wasted_bits_buffer[0][i]; + a = (a << wasted_bits) | wasted_bits_buffer[1][i]; + } + + buffer_out[i * numchannels] = b << 8; + buffer_out[i * numchannels + 1] = a << 8; + } + } else { + for (i = 0; i < numsamples; i++) { + int32_t left, right; + + left = buffer[0][i]; + right = buffer[1][i]; + + if (wasted_bits) { + left = (left << wasted_bits) | wasted_bits_buffer[0][i]; + right = (right << wasted_bits) | wasted_bits_buffer[1][i]; + } + + buffer_out[i * numchannels] = left << 8; + buffer_out[i * numchannels + 1] = right << 8; + } + } +} + static int alac_decode_frame(AVCodecContext *avctx, void *outbuffer, int *outputsize, AVPacket *avpkt) @@ -410,7 +465,6 @@ unsigned int outputsamples; int hassize; unsigned int readsamplesize; - int wasted_bytes; int isnotcompressed; uint8_t interlacing_shift; uint8_t interlacing_leftweight; @@ -452,7 +506,7 @@ /* the output sample size is stored soon */ hassize = get_bits1(&alac->gb); - wasted_bytes = get_bits(&alac->gb, 2); /* unknown ? */ + alac->wasted_bits = get_bits(&alac->gb, 2) << 3; /* whether the frame is compressed */ isnotcompressed = get_bits1(&alac->gb); @@ -467,13 +521,25 @@ } else outputsamples = alac->setinfo_max_samples_per_frame; + switch (alac->setinfo_sample_size) { + case 16: avctx->sample_fmt = SAMPLE_FMT_S16; + alac->bytespersample = channels << 1; + break; + case 24: avctx->sample_fmt = SAMPLE_FMT_S32; + alac->bytespersample = channels << 2; + break; + default: av_log(avctx, AV_LOG_ERROR, "Sample depth %d is not supported.\n", + alac->setinfo_sample_size); + return -1; + } + if(outputsamples > *outputsize / alac->bytespersample){ av_log(avctx, AV_LOG_ERROR, "sample buffer too small\n"); return -1; } *outputsize = outputsamples * alac->bytespersample; - readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8) + channels - 1; + readsamplesize = alac->setinfo_sample_size - (alac->wasted_bits) + channels - 1; if (readsamplesize > MIN_CACHE_BITS) { av_log(avctx, AV_LOG_ERROR, "readsamplesize too big (%d)\n", readsamplesize); return -1; @@ -503,9 +569,13 @@ predictor_coef_table[chan][i] = (int16_t)get_bits(&alac->gb, 16); } - if (wasted_bytes) - av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented, unhandling of wasted_bytes\n"); - + if (alac->wasted_bits) { + int i, ch; + for (i = 0; i < outputsamples; i++) { + for (ch = 0; ch < channels; ch++) + alac->wasted_bits_buffer[ch][i] = get_bits(&alac->gb, alac->wasted_bits); + } + } for (chan = 0; chan < channels; chan++) { bastardized_rice_decompress(alac, alac->predicterror_buffer[chan], @@ -538,6 +608,7 @@ } else { /* not compressed, easy case */ int i, chan; + if (alac->setinfo_sample_size <= 16) { for (i = 0; i < outputsamples; i++) for (chan = 0; chan < channels; chan++) { int32_t audiobits; @@ -546,7 +617,17 @@ alac->outputsamples_buffer[chan][i] = audiobits; } - /* wasted_bytes = 0; */ + } else { + for (i = 0; i < outputsamples; i++) { + for (chan = 0; chan < channels; chan++) { + alac->outputsamples_buffer[chan][i] = get_bits(&alac->gb, + alac->setinfo_sample_size); + alac->outputsamples_buffer[chan][i] = sign_extend(alac->outputsamples_buffer[chan][i], + alac->setinfo_sample_size); + } + } + } + alac->wasted_bits = 0; interlacing_shift = 0; interlacing_leftweight = 0; } @@ -570,14 +651,21 @@ } } break; - case 20: case 24: - // It is not clear if there exist any encoder that creates 24 bit ALAC - // files. iTunes convert 24 bit raw files to 16 bit before encoding. - case 32: - av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented sample size %i\n", alac->setinfo_sample_size); - break; - default: + if (channels == 2) { + decorrelate_stereo_24(alac->outputsamples_buffer, + outbuffer, + alac->wasted_bits_buffer, + alac->wasted_bits, + alac->numchannels, + outputsamples, + interlacing_shift, + interlacing_leftweight); + } else { + int i; + for (i = 0; i < outputsamples; i++) + ((int32_t *)outbuffer)[i] = alac->outputsamples_buffer[0][i] << 8; + } break; } @@ -594,8 +682,6 @@ alac->context_initialized = 0; alac->numchannels = alac->avctx->channels; - alac->bytespersample = 2 * alac->numchannels; - avctx->sample_fmt = SAMPLE_FMT_S16; return 0; } @@ -608,6 +694,7 @@ for (chan = 0; chan < MAX_CHANNELS; chan++) { av_free(alac->predicterror_buffer[chan]); av_free(alac->outputsamples_buffer[chan]); + av_freep(&alac->wasted_bits_buffer[chan]); } return 0;