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downmix before imdct unless different size transforms are used. about 20%
faster 5.1-to-stereo downmixing. Originally committed as revision 12397 to svn://svn.ffmpeg.org/ffmpeg/trunk
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@ -171,6 +171,7 @@ typedef struct {
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int fixed_coeffs[AC3_MAX_CHANNELS][256]; ///> fixed-point transform coefficients
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DECLARE_ALIGNED_16(float, transform_coeffs[AC3_MAX_CHANNELS][256]); ///< transform coefficients
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int downmixed; ///< indicates if coeffs are currently downmixed
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/* For IMDCT. */
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MDCTContext imdct_512; ///< for 512 sample IMDCT
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@ -179,9 +180,9 @@ typedef struct {
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float add_bias; ///< offset for float_to_int16 conversion
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float mul_bias; ///< scaling for float_to_int16 conversion
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DECLARE_ALIGNED_16(float, output[AC3_MAX_CHANNELS-1][256]); ///< output after imdct transform and windowing
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DECLARE_ALIGNED_16(float, output[AC3_MAX_CHANNELS][256]); ///< output after imdct transform and windowing
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DECLARE_ALIGNED_16(short, int_output[AC3_MAX_CHANNELS-1][256]); ///< final 16-bit integer output
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DECLARE_ALIGNED_16(float, delay[AC3_MAX_CHANNELS-1][256]); ///< delay - added to the next block
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DECLARE_ALIGNED_16(float, delay[AC3_MAX_CHANNELS][256]); ///< delay - added to the next block
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DECLARE_ALIGNED_16(float, tmp_imdct[256]); ///< temporary storage for imdct transform
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DECLARE_ALIGNED_16(float, tmp_output[512]); ///< temporary storage for output before windowing
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DECLARE_ALIGNED_16(float, window[256]); ///< window coefficients
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@ -287,6 +288,7 @@ static int ac3_decode_init(AVCodecContext *avctx)
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avctx->request_channels <= 2) {
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avctx->channels = avctx->request_channels;
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}
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s->downmixed = 1;
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return 0;
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}
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@ -708,15 +710,9 @@ static void do_imdct_256(AC3DecodeContext *s, int chindex)
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* Convert frequency domain coefficients to time-domain audio samples.
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* reference: Section 7.9.4 Transformation Equations
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*/
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static inline void do_imdct(AC3DecodeContext *s)
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static inline void do_imdct(AC3DecodeContext *s, int channels)
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{
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int ch;
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int channels;
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/* Don't perform the IMDCT on the LFE channel unless it's used in the output */
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channels = s->fbw_channels;
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if(s->output_mode & AC3_OUTPUT_LFEON)
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channels++;
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for (ch=1; ch<=channels; ch++) {
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if (s->block_switch[ch]) {
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@ -739,7 +735,8 @@ static inline void do_imdct(AC3DecodeContext *s)
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/**
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* Downmix the output to mono or stereo.
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*/
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static void ac3_downmix(AC3DecodeContext *s)
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static void ac3_downmix(AC3DecodeContext *s,
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float samples[AC3_MAX_CHANNELS][256], int ch_offset)
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{
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int i, j;
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float v0, v1;
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@ -747,20 +744,48 @@ static void ac3_downmix(AC3DecodeContext *s)
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for(i=0; i<256; i++) {
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v0 = v1 = 0.0f;
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for(j=0; j<s->fbw_channels; j++) {
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v0 += s->output[j][i] * s->downmix_coeffs[j][0];
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v1 += s->output[j][i] * s->downmix_coeffs[j][1];
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v0 += samples[j+ch_offset][i] * s->downmix_coeffs[j][0];
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v1 += samples[j+ch_offset][i] * s->downmix_coeffs[j][1];
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}
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v0 *= s->downmix_coeff_adjust[0];
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v1 *= s->downmix_coeff_adjust[1];
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if(s->output_mode == AC3_CHMODE_MONO) {
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s->output[0][i] = (v0 + v1) * LEVEL_MINUS_3DB;
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samples[ch_offset][i] = (v0 + v1) * LEVEL_MINUS_3DB;
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} else if(s->output_mode == AC3_CHMODE_STEREO) {
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s->output[0][i] = v0;
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s->output[1][i] = v1;
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samples[ ch_offset][i] = v0;
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samples[1+ch_offset][i] = v1;
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}
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}
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}
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/**
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* Upmix delay samples from stereo to original channel layout.
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*/
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static void ac3_upmix_delay(AC3DecodeContext *s)
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{
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int channel_data_size = sizeof(s->delay[0]);
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switch(s->channel_mode) {
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case AC3_CHMODE_DUALMONO:
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case AC3_CHMODE_STEREO:
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/* upmix mono to stereo */
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memcpy(s->delay[1], s->delay[0], channel_data_size);
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break;
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case AC3_CHMODE_2F2R:
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memset(s->delay[3], 0, channel_data_size);
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case AC3_CHMODE_2F1R:
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memset(s->delay[2], 0, channel_data_size);
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break;
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case AC3_CHMODE_3F2R:
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memset(s->delay[4], 0, channel_data_size);
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case AC3_CHMODE_3F1R:
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memset(s->delay[3], 0, channel_data_size);
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case AC3_CHMODE_3F:
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memcpy(s->delay[2], s->delay[1], channel_data_size);
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memset(s->delay[1], 0, channel_data_size);
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break;
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}
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}
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/**
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* Parse an audio block from AC-3 bitstream.
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*/
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@ -769,14 +794,20 @@ static int ac3_parse_audio_block(AC3DecodeContext *s, int blk)
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int fbw_channels = s->fbw_channels;
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int channel_mode = s->channel_mode;
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int i, bnd, seg, ch;
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int different_transforms;
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int downmix_output;
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GetBitContext *gbc = &s->gbc;
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uint8_t bit_alloc_stages[AC3_MAX_CHANNELS];
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memset(bit_alloc_stages, 0, AC3_MAX_CHANNELS);
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/* block switch flags */
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for (ch = 1; ch <= fbw_channels; ch++)
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different_transforms = 0;
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for (ch = 1; ch <= fbw_channels; ch++) {
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s->block_switch[ch] = get_bits1(gbc);
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if(ch > 1 && s->block_switch[ch] != s->block_switch[1])
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different_transforms = 1;
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}
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/* dithering flags */
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s->dither_all = 1;
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@ -1048,12 +1079,36 @@ static int ac3_parse_audio_block(AC3DecodeContext *s, int blk)
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}
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}
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do_imdct(s);
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/* downmix and MDCT. order depends on whether block switching is used for
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any channel in this block. this is because coefficients for the long
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and short transforms cannot be mixed. */
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downmix_output = s->channels != s->out_channels &&
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!((s->output_mode & AC3_OUTPUT_LFEON) &&
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s->fbw_channels == s->out_channels);
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if(different_transforms) {
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/* the delay samples have already been downmixed, so we upmix the delay
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samples in order to reconstruct all channels before downmixing. */
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if(s->downmixed) {
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s->downmixed = 0;
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ac3_upmix_delay(s);
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}
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/* downmix output if needed */
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if(s->channels != s->out_channels && !((s->output_mode & AC3_OUTPUT_LFEON) &&
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s->fbw_channels == s->out_channels)) {
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ac3_downmix(s);
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do_imdct(s, s->channels);
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if(downmix_output) {
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ac3_downmix(s, s->output, 0);
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}
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} else {
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if(downmix_output) {
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ac3_downmix(s, s->transform_coeffs, 1);
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}
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if(!s->downmixed) {
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s->downmixed = 1;
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ac3_downmix(s, s->delay, 0);
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}
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do_imdct(s, s->out_channels);
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}
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/* convert float to 16-bit integer */
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