diff options
| -rw-r--r-- | libavcodec/ac3dec.c | 97 |
1 files changed, 76 insertions, 21 deletions
diff --git a/libavcodec/ac3dec.c b/libavcodec/ac3dec.c index 35a8ed1f25..1c8bec8850 100644 --- a/libavcodec/ac3dec.c +++ b/libavcodec/ac3dec.c @@ -171,6 +171,7 @@ typedef struct { int fixed_coeffs[AC3_MAX_CHANNELS][256]; ///> fixed-point transform coefficients DECLARE_ALIGNED_16(float, transform_coeffs[AC3_MAX_CHANNELS][256]); ///< transform coefficients + int downmixed; ///< indicates if coeffs are currently downmixed /* For IMDCT. */ MDCTContext imdct_512; ///< for 512 sample IMDCT @@ -179,9 +180,9 @@ typedef struct { float add_bias; ///< offset for float_to_int16 conversion float mul_bias; ///< scaling for float_to_int16 conversion - DECLARE_ALIGNED_16(float, output[AC3_MAX_CHANNELS-1][256]); ///< output after imdct transform and windowing + DECLARE_ALIGNED_16(float, output[AC3_MAX_CHANNELS][256]); ///< output after imdct transform and windowing DECLARE_ALIGNED_16(short, int_output[AC3_MAX_CHANNELS-1][256]); ///< final 16-bit integer output - DECLARE_ALIGNED_16(float, delay[AC3_MAX_CHANNELS-1][256]); ///< delay - added to the next block + DECLARE_ALIGNED_16(float, delay[AC3_MAX_CHANNELS][256]); ///< delay - added to the next block DECLARE_ALIGNED_16(float, tmp_imdct[256]); ///< temporary storage for imdct transform DECLARE_ALIGNED_16(float, tmp_output[512]); ///< temporary storage for output before windowing DECLARE_ALIGNED_16(float, window[256]); ///< window coefficients @@ -287,6 +288,7 @@ static int ac3_decode_init(AVCodecContext *avctx) avctx->request_channels <= 2) { avctx->channels = avctx->request_channels; } + s->downmixed = 1; return 0; } @@ -708,15 +710,9 @@ static void do_imdct_256(AC3DecodeContext *s, int chindex) * Convert frequency domain coefficients to time-domain audio samples. * reference: Section 7.9.4 Transformation Equations */ -static inline void do_imdct(AC3DecodeContext *s) +static inline void do_imdct(AC3DecodeContext *s, int channels) { int ch; - int channels; - - /* Don't perform the IMDCT on the LFE channel unless it's used in the output */ - channels = s->fbw_channels; - if(s->output_mode & AC3_OUTPUT_LFEON) - channels++; for (ch=1; ch<=channels; ch++) { if (s->block_switch[ch]) { @@ -739,7 +735,8 @@ static inline void do_imdct(AC3DecodeContext *s) /** * Downmix the output to mono or stereo. */ -static void ac3_downmix(AC3DecodeContext *s) +static void ac3_downmix(AC3DecodeContext *s, + float samples[AC3_MAX_CHANNELS][256], int ch_offset) { int i, j; float v0, v1; @@ -747,21 +744,49 @@ static void ac3_downmix(AC3DecodeContext *s) for(i=0; i<256; i++) { v0 = v1 = 0.0f; for(j=0; j<s->fbw_channels; j++) { - v0 += s->output[j][i] * s->downmix_coeffs[j][0]; - v1 += s->output[j][i] * s->downmix_coeffs[j][1]; + v0 += samples[j+ch_offset][i] * s->downmix_coeffs[j][0]; + v1 += samples[j+ch_offset][i] * s->downmix_coeffs[j][1]; } v0 *= s->downmix_coeff_adjust[0]; v1 *= s->downmix_coeff_adjust[1]; if(s->output_mode == AC3_CHMODE_MONO) { - s->output[0][i] = (v0 + v1) * LEVEL_MINUS_3DB; + samples[ch_offset][i] = (v0 + v1) * LEVEL_MINUS_3DB; } else if(s->output_mode == AC3_CHMODE_STEREO) { - s->output[0][i] = v0; - s->output[1][i] = v1; + samples[ ch_offset][i] = v0; + samples[1+ch_offset][i] = v1; } } } /** + * Upmix delay samples from stereo to original channel layout. + */ +static void ac3_upmix_delay(AC3DecodeContext *s) +{ + int channel_data_size = sizeof(s->delay[0]); + switch(s->channel_mode) { + case AC3_CHMODE_DUALMONO: + case AC3_CHMODE_STEREO: + /* upmix mono to stereo */ + memcpy(s->delay[1], s->delay[0], channel_data_size); + break; + case AC3_CHMODE_2F2R: + memset(s->delay[3], 0, channel_data_size); + case AC3_CHMODE_2F1R: + memset(s->delay[2], 0, channel_data_size); + break; + case AC3_CHMODE_3F2R: + memset(s->delay[4], 0, channel_data_size); + case AC3_CHMODE_3F1R: + memset(s->delay[3], 0, channel_data_size); + case AC3_CHMODE_3F: + memcpy(s->delay[2], s->delay[1], channel_data_size); + memset(s->delay[1], 0, channel_data_size); + break; + } +} + +/** * Parse an audio block from AC-3 bitstream. */ static int ac3_parse_audio_block(AC3DecodeContext *s, int blk) @@ -769,14 +794,20 @@ static int ac3_parse_audio_block(AC3DecodeContext *s, int blk) int fbw_channels = s->fbw_channels; int channel_mode = s->channel_mode; int i, bnd, seg, ch; + int different_transforms; + int downmix_output; GetBitContext *gbc = &s->gbc; uint8_t bit_alloc_stages[AC3_MAX_CHANNELS]; memset(bit_alloc_stages, 0, AC3_MAX_CHANNELS); /* block switch flags */ - for (ch = 1; ch <= fbw_channels; ch++) + different_transforms = 0; + for (ch = 1; ch <= fbw_channels; ch++) { s->block_switch[ch] = get_bits1(gbc); + if(ch > 1 && s->block_switch[ch] != s->block_switch[1]) + different_transforms = 1; + } /* dithering flags */ s->dither_all = 1; @@ -1048,12 +1079,36 @@ static int ac3_parse_audio_block(AC3DecodeContext *s, int blk) } } - do_imdct(s); + /* downmix and MDCT. order depends on whether block switching is used for + any channel in this block. this is because coefficients for the long + and short transforms cannot be mixed. */ + downmix_output = s->channels != s->out_channels && + !((s->output_mode & AC3_OUTPUT_LFEON) && + s->fbw_channels == s->out_channels); + if(different_transforms) { + /* the delay samples have already been downmixed, so we upmix the delay + samples in order to reconstruct all channels before downmixing. */ + if(s->downmixed) { + s->downmixed = 0; + ac3_upmix_delay(s); + } - /* downmix output if needed */ - if(s->channels != s->out_channels && !((s->output_mode & AC3_OUTPUT_LFEON) && - s->fbw_channels == s->out_channels)) { - ac3_downmix(s); + do_imdct(s, s->channels); + + if(downmix_output) { + ac3_downmix(s, s->output, 0); + } + } else { + if(downmix_output) { + ac3_downmix(s, s->transform_coeffs, 1); + } + + if(!s->downmixed) { + s->downmixed = 1; + ac3_downmix(s, s->delay, 0); + } + + do_imdct(s, s->out_channels); } /* convert float to 16-bit integer */ |