July 2009 - FFmpeg-soc - lists.ffmpeg.org

[soc]: r4813 - amr/amrnbfloatdec.c
by cmcq 28 Jul '09

28 Jul '09

Author: cmcq Date: Tue Jul 28 17:33:24 2009 New Revision: 4813 Log: Rename post_process to postfilter; post_process is a misnomer Modified: amr/amrnbfloatdec.c Modified: amr/amrnbfloatdec.c ============================================================================== --- amr/amrnbfloatdec.c Tue Jul 28 17:28:31 2009 (r4812) +++ amr/amrnbfloatdec.c Tue Jul 28 17:33:24 2009 (r4813) @@ -78,9 +78,9 @@ typedef struct AMRContext { uint8_t prev_ir_filter_strength; ///< previous impulse response filter strength; 0 - strong, 1 - medium, 2 - none uint8_t ir_filter_onset; ///< flag for impulse response filter strength - float post_process_mem[10]; ///< previous intermediate values in the formant filter + float postfilter_mem[10]; ///< previous intermediate values in the formant filter float tilt_mem; ///< previous input to tilt compensation filter - float post_filter_agc; ///< previous factor used for adaptive gain control + float postfilter_agc; ///< previous factor used for adaptive gain control float high_pass_mem[2]; ///< previous intermediate values in the high-pass filter float samples_in[LP_FILTER_ORDER + AMR_SUBFRAME_SIZE]; ///< floating point samples @@ -1079,13 +1079,13 @@ static void tilt_compensation(float *mem * @param lpc interpolated LP coefficients for this subframe * @param buf_out output of the filter */ -static void post_process(AMRContext *p, float *lpc, float *buf_out) +static void postfilter(AMRContext *p, float *lpc, float *buf_out) { int i; float *samples = p->samples_in + LP_FILTER_ORDER; float gain_scale_factor = 1.0; float speech_gain = ff_dot_productf(samples, samples, AMR_SUBFRAME_SIZE); - float post_filter_gain; + float postfilter_gain; float tmp[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER]; const float *gamma_n, *gamma_d; // Formant filter factor table float lpc_n[LP_FILTER_ORDER + 1], // Transfer function coefficients @@ -1106,10 +1106,10 @@ static void post_process(AMRContext *p, } // Apply transfer function given by lpc_n and lpc_d - memcpy(tmp, p->post_process_mem, sizeof(float) * LP_FILTER_ORDER); + memcpy(tmp, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER); ff_celp_lp_synthesis_filterf(tmp + LP_FILTER_ORDER, lpc_d, samples, AMR_SUBFRAME_SIZE, LP_FILTER_ORDER); - memcpy(p->post_process_mem, tmp + AMR_SUBFRAME_SIZE, + memcpy(p->postfilter_mem, tmp + AMR_SUBFRAME_SIZE, sizeof(float) * LP_FILTER_ORDER); ff_celp_lp_zero_synthesis_filterf(buf_out, lpc_n, tmp + LP_FILTER_ORDER, AMR_SUBFRAME_SIZE, LP_FILTER_ORDER + 1); @@ -1118,14 +1118,14 @@ static void post_process(AMRContext *p, tilt_compensation(&p->tilt_mem, tilt_factor(lpc_n, lpc_d), buf_out); // Adaptive gain control - post_filter_gain = ff_dot_productf(buf_out, buf_out, AMR_SUBFRAME_SIZE); - if (post_filter_gain) - gain_scale_factor = sqrt(speech_gain / post_filter_gain); + postfilter_gain = ff_dot_productf(buf_out, buf_out, AMR_SUBFRAME_SIZE); + if (postfilter_gain) + gain_scale_factor = sqrt(speech_gain / postfilter_gain); for (i = 0; i < AMR_SUBFRAME_SIZE; i++) { - p->post_filter_agc = AMR_AGC_ALPHA * p->post_filter_agc + + p->postfilter_agc = AMR_AGC_ALPHA * p->postfilter_agc + (1.0 - AMR_AGC_ALPHA) * gain_scale_factor; - buf_out[i] *= p->post_filter_agc; + buf_out[i] *= p->postfilter_agc; } } @@ -1247,7 +1247,7 @@ static int amrnb_decode_frame(AVCodecCon /*** end of synthesis ***/ - post_process(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE); + postfilter(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE); high_pass_filter(p->high_pass_mem, buf_out + subframe * AMR_SUBFRAME_SIZE); // update buffers and history

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[soc]: r4812 - amr/amrnbfloatdec.c
by cmcq 28 Jul '09

28 Jul '09

Author: cmcq Date: Tue Jul 28 17:28:31 2009 New Revision: 4812 Log: Simplify (x != 0) ==> (x) Modified: amr/amrnbfloatdec.c Modified: amr/amrnbfloatdec.c ============================================================================== --- amr/amrnbfloatdec.c Mon Jul 27 13:27:32 2009 (r4811) +++ amr/amrnbfloatdec.c Tue Jul 28 17:28:31 2009 (r4812) @@ -1119,7 +1119,7 @@ static void post_process(AMRContext *p, // Adaptive gain control post_filter_gain = ff_dot_productf(buf_out, buf_out, AMR_SUBFRAME_SIZE); - if (post_filter_gain != 0) + if (post_filter_gain) gain_scale_factor = sqrt(speech_gain / post_filter_gain); for (i = 0; i < AMR_SUBFRAME_SIZE; i++) {

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[PATCH] Some fast_bilinear patches.
by Ramiro Polla 28 Jul '09

28 Jul '09

Hi, Attached are some patches for the fast bilinear scaler, while I wait on Michael's answer on the previous message regarding fast_bilinear. Hmm, should these patches go to -devel or -soc? Ramiro Polla

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[soc]: r4811 - als/alsdec.c
by thilo.borgmann 27 Jul '09

27 Jul '09

Author: thilo.borgmann Date: Mon Jul 27 13:27:32 2009 New Revision: 4811 Log: Corrected spelling in some comments. Modified: als/alsdec.c Modified: als/alsdec.c ============================================================================== --- als/alsdec.c Sun Jul 26 14:20:09 2009 (r4810) +++ als/alsdec.c Mon Jul 27 13:27:32 2009 (r4811) @@ -47,7 +47,7 @@ typedef struct { int random_access; ///< Distance between RA frames (in frames, 0...255) int ra_flag; ///< Indicates where the size of ra units is stored. int adapt_order; ///< Adaptive order: 1 = on, 0 = off - int coef_table; ///< Table index of rice code parameters + int coef_table; ///< Table index of Rice code parameters int long_term_prediction; ///< Long term prediction (LTP): 1 = on, 0 = off int max_order; ///< Maximum prediction order (0..1023) int block_switching; ///< Number of block switching levels @@ -296,7 +296,7 @@ static void parse_bs_info(uint32_t bs_in } -/** Reads and decodes a rice codeword. +/** Reads and decodes a Rice codeword. */ static int64_t decode_rice(GetBitContext *gb, unsigned int k) {

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[soc]: r4810 - wmapro/wmaprodec.c
by faust3 26 Jul '09

26 Jul '09

Author: faust3 Date: Sun Jul 26 14:20:09 2009 New Revision: 4810 Log: remove redundant special casing Modified: wmapro/wmaprodec.c Modified: wmapro/wmaprodec.c ============================================================================== --- wmapro/wmaprodec.c Sun Jul 26 14:15:19 2009 (r4809) +++ wmapro/wmaprodec.c Sun Jul 26 14:20:09 2009 (r4810) @@ -1119,6 +1119,7 @@ static int decode_subframe(WMA3DecodeCon int i; int total_samples = s->samples_per_frame * s->num_channels; int transmit_coeffs = 0; + int frame_offset; s->subframe_offset = get_bits_count(&s->gb); @@ -1166,16 +1167,10 @@ static int decode_subframe(WMA3DecodeCon s->channels_for_cur_subframe); /** calculate number of scale factor bands and their offsets */ - if (subframe_len == s->samples_per_frame) { - s->num_bands = s->num_sfb[0]; - s->cur_sfb_offsets = s->sfb_offsets; - s->cur_subwoofer_cutoff = s->subwoofer_cutoffs[0]; - } else { - int frame_offset = av_log2(s->samples_per_frame/subframe_len); - s->num_bands = s->num_sfb[frame_offset]; - s->cur_sfb_offsets = &s->sfb_offsets[MAX_BANDS * frame_offset]; - s->cur_subwoofer_cutoff = s->subwoofer_cutoffs[frame_offset]; - } + frame_offset = av_log2(s->samples_per_frame/subframe_len); + s->num_bands = s->num_sfb[frame_offset]; + s->cur_sfb_offsets = &s->sfb_offsets[MAX_BANDS * frame_offset]; + s->cur_subwoofer_cutoff = s->subwoofer_cutoffs[frame_offset]; /** configure the decoder for the current subframe */ for (i=0;i<s->channels_for_cur_subframe;i++) {

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[soc]: r4809 - wmapro/wmaprodec.c
by faust3 26 Jul '09

26 Jul '09

Author: faust3 Date: Sun Jul 26 14:15:19 2009 New Revision: 4809 Log: prefix window function with wmapro Modified: wmapro/wmaprodec.c Modified: wmapro/wmaprodec.c ============================================================================== --- wmapro/wmaprodec.c Sun Jul 26 14:08:47 2009 (r4808) +++ wmapro/wmaprodec.c Sun Jul 26 14:15:19 2009 (r4809) @@ -1082,7 +1082,7 @@ static void inverse_channel_transform(WM *@brief Apply sine window and reconstruct the output buffer. *@param s codec context */ -static void window(WMA3DecodeContext *s) +static void wmapro_window(WMA3DecodeContext *s) { int i; for (i=0;i<s->channels_for_cur_subframe;i++) { @@ -1334,7 +1334,7 @@ static int decode_subframe(WMA3DecodeCon } /** window and overlapp-add */ - window(s); + wmapro_window(s); /** handled one subframe */ for (i=0;i<s->channels_for_cur_subframe;i++) {

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[soc]: r4808 - wmapro/wmaprodec.c
by faust3 26 Jul '09

26 Jul '09

Author: faust3 Date: Sun Jul 26 14:08:47 2009 New Revision: 4808 Log: remove unneeded shift Modified: wmapro/wmaprodec.c Modified: wmapro/wmaprodec.c ============================================================================== --- wmapro/wmaprodec.c Sun Jul 26 13:49:32 2009 (r4807) +++ wmapro/wmaprodec.c Sun Jul 26 14:08:47 2009 (r4808) @@ -301,7 +301,7 @@ static av_cold int decode_init(AVCodecCo /** frame info */ s->skip_frame = 1; /** skip first frame */ s->packet_loss = 1; - s->len_prefix = (s->decode_flags & 0x40) >> 6; + s->len_prefix = (s->decode_flags & 0x40); if (!s->len_prefix) { av_log_ask_for_sample(avctx, "no length prefix\n");

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[soc]: r4807 - wmapro/wmaprodec.c
by faust3 26 Jul '09

26 Jul '09

Author: faust3 Date: Sun Jul 26 13:49:32 2009 New Revision: 4807 Log: clarify some comments Modified: wmapro/wmaprodec.c Modified: wmapro/wmaprodec.c ============================================================================== --- wmapro/wmaprodec.c Fri Jul 24 12:32:00 2009 (r4806) +++ wmapro/wmaprodec.c Sun Jul 26 13:49:32 2009 (r4807) @@ -44,8 +44,20 @@ * subframes. Every subframe contains the data for 2^N time domain samples * where N varies between 7 and 12. * + * Example wmapro bitstream (in samples): + * + * || packet 0 || packet 1 || packet 2 packets + * --------------------------------------------------- + * || frame 0 || frame 1 || frame 2 || frames + * --------------------------------------------------- + * || | | || | | | || || subframes of channel 0 + * --------------------------------------------------- + * || | | || | | | || || subframes of channel 1 + * --------------------------------------------------- + * * The frame layouts for the individual channels of a wma frame does not need * to be the same. + * * However, if the offsets and lengths of several subframes of a frame are the * same, the subframes of the channels can be grouped. * Every group may then use special coding techniques like M/S stereo coding @@ -109,27 +121,27 @@ static VLC coef_vlc[2]; static float sin64[33]; ///< sinus table for decorrelation /** - * @brief decoder context for a single channel + * @brief frame specific decoder context for a single channel */ typedef struct { int16_t prev_block_len; ///< length of the previous block uint8_t transmit_coefs; uint8_t num_subframes; uint16_t subframe_len[MAX_SUBFRAMES]; ///< subframe length in samples - uint16_t subframe_offset[MAX_SUBFRAMES]; ///< subframe position - uint8_t cur_subframe; ///< subframe index - uint16_t channel_len; ///< channel length in samples + uint16_t subframe_offset[MAX_SUBFRAMES]; ///< subframe positions in the current frame + uint8_t cur_subframe; ///< current subframe number + uint16_t channel_len; ///< channel frame length in samples uint16_t decoded_samples; ///< already processed samples uint8_t grouped; ///< channel is part of a group - int quant_step; ///< quantization step - int8_t transmit_sf; ///< transmit scale factors + int quant_step; ///< quantization step for the current subframe + int8_t transmit_sf; ///< transmit scale factors for the current subframe int8_t reuse_sf; ///< share scale factors between subframes - int8_t scale_factor_step; ///< scaling step - int max_scale_factor; ///< maximum scale factor - int scale_factors[MAX_BANDS]; ///< scale factor values - int resampled_scale_factors[MAX_BANDS]; ///< scale factors from a previous block + int8_t scale_factor_step; ///< scaling step for the current subframe + int max_scale_factor; ///< maximum scale factor for the current subframe + int scale_factors[MAX_BANDS]; ///< scale factor values for the current subframe + int resampled_scale_factors[MAX_BANDS]; ///< scale factors from a previous subframe int16_t scale_factor_block_len; ///< scale factor reference block length - float* coeffs; ///< pointer to the decode buffer + float* coeffs; ///< pointer to the subframe decode buffer DECLARE_ALIGNED_16(float, out[2*WMAPRO_BLOCK_MAX_SIZE]); ///< output buffer } WMA3ChannelCtx; @@ -465,7 +477,7 @@ static av_cold int decode_init(AVCodecCo * If the subframes are not evenly split, the algorithm estimates the * channels with the lowest number of total samples. * Afterwards, for each of these channels a bit is read from the - * bitstream that indicates if the channel contains a frame with the + * bitstream that indicates if the channel contains a subframe with the * next subframe size that is going to be read from the bitstream or not. * If a channel contains such a subframe, the subframe size gets added to * the channel's subframe list.

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Differing
by Ramiro Polla 24 Jul '09

24 Jul '09

Hi, In swscale.c, there are some lines that adjust {lum,chr}XInc depending on MMX or MMX2 being used for fast bilinear. They were added (still with the old name of s_xinc) in this commit: http://git.ffmpeg.org/?p=libswscale;a=commitdiff;h=87d2542e33c80731dab64339… Where do the magical "20" values come from? >From what I understood and from some tests I've done, this creates some rather ugly last lines at the far right of the image. The output also differs for C, mmx, and mmx2, and it's not under BITEXACT. To remove this adjustment and have the same output for all configurations, wouldn't it only be necessary to: - adjust mmx and mmx2 filters to only work up to the last input pixels it can before reading past the end. - run the C code on the last pixels if necessary. This should also remove some +1s being allocated because of the MMX filter, and the limit that the MMX2 filter can only work with srcW&31. Ramiro Polla

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please remove me from this list
by Divyesh Dabhi 24 Jul '09

24 Jul '09

please remove dd(a)ji-tu.com from this list thank you

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