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Sun, 08 Dec 2019 07:11:43 -0800 (PST) Received: from localhost.localdomain ([181.23.68.111]) by smtp.gmail.com with ESMTPSA id b185sm3528055vke.22.2019.12.08.07.11.41 for (version=TLS1_3 cipher=TLS_AES_256_GCM_SHA384 bits=256/256); Sun, 08 Dec 2019 07:11:43 -0800 (PST) From: James Almer To: ffmpeg-devel@ffmpeg.org Date: Sun, 8 Dec 2019 12:11:28 -0300 Message-Id: <20191208151128.2169-1-jamrial@gmail.com> X-Mailer: git-send-email 2.24.0 MIME-Version: 1.0 Subject: [FFmpeg-devel] [PATCH] avcodec/aptx: split decoder and encoder into separate files X-BeenThere: ffmpeg-devel@ffmpeg.org X-Mailman-Version: 2.1.20 Precedence: list List-Id: FFmpeg development discussions and patches List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Reply-To: FFmpeg development discussions and patches Errors-To: ffmpeg-devel-bounces@ffmpeg.org Sender: "ffmpeg-devel" Signed-off-by: James Almer --- Untested beyond checking it compiles because there are no FATE tests for either module. libavcodec/Makefile | 8 +- libavcodec/aptx.c | 634 +------------------------------------------ libavcodec/aptx.h | 221 +++++++++++++++ libavcodec/aptxdec.c | 206 ++++++++++++++ libavcodec/aptxenc.c | 271 ++++++++++++++++++ 5 files changed, 709 insertions(+), 631 deletions(-) create mode 100644 libavcodec/aptx.h create mode 100644 libavcodec/aptxdec.c create mode 100644 libavcodec/aptxenc.c diff --git a/libavcodec/Makefile b/libavcodec/Makefile index c1f35b40d8..0d933a8fb9 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -194,10 +194,10 @@ OBJS-$(CONFIG_AMV_ENCODER) += mjpegenc.o mjpegenc_common.o \ OBJS-$(CONFIG_ANM_DECODER) += anm.o OBJS-$(CONFIG_ANSI_DECODER) += ansi.o cga_data.o OBJS-$(CONFIG_APE_DECODER) += apedec.o -OBJS-$(CONFIG_APTX_DECODER) += aptx.o -OBJS-$(CONFIG_APTX_ENCODER) += aptx.o -OBJS-$(CONFIG_APTX_HD_DECODER) += aptx.o -OBJS-$(CONFIG_APTX_HD_ENCODER) += aptx.o +OBJS-$(CONFIG_APTX_DECODER) += aptxdec.o aptx.o +OBJS-$(CONFIG_APTX_ENCODER) += aptxenc.o aptx.o +OBJS-$(CONFIG_APTX_HD_DECODER) += aptxdec.o aptx.o +OBJS-$(CONFIG_APTX_HD_ENCODER) += aptxenc.o aptx.o OBJS-$(CONFIG_APNG_DECODER) += png.o pngdec.o pngdsp.o OBJS-$(CONFIG_APNG_ENCODER) += png.o pngenc.o OBJS-$(CONFIG_ARBC_DECODER) += arbc.o diff --git a/libavcodec/aptx.c b/libavcodec/aptx.c index a2620a9212..c7ee45aaa1 100644 --- a/libavcodec/aptx.c +++ b/libavcodec/aptx.c @@ -20,81 +20,7 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#include "libavutil/intreadwrite.h" -#include "avcodec.h" -#include "internal.h" -#include "mathops.h" -#include "audio_frame_queue.h" - - -enum channels { - LEFT, - RIGHT, - NB_CHANNELS -}; - -enum subbands { - LF, // Low Frequency (0-5.5 kHz) - MLF, // Medium-Low Frequency (5.5-11kHz) - MHF, // Medium-High Frequency (11-16.5kHz) - HF, // High Frequency (16.5-22kHz) - NB_SUBBANDS -}; - -#define NB_FILTERS 2 -#define FILTER_TAPS 16 - -typedef struct { - int pos; - int32_t buffer[2*FILTER_TAPS]; -} FilterSignal; - -typedef struct { - FilterSignal outer_filter_signal[NB_FILTERS]; - FilterSignal inner_filter_signal[NB_FILTERS][NB_FILTERS]; -} QMFAnalysis; - -typedef struct { - int32_t quantized_sample; - int32_t quantized_sample_parity_change; - int32_t error; -} Quantize; - -typedef struct { - int32_t quantization_factor; - int32_t factor_select; - int32_t reconstructed_difference; -} InvertQuantize; - -typedef struct { - int32_t prev_sign[2]; - int32_t s_weight[2]; - int32_t d_weight[24]; - int32_t pos; - int32_t reconstructed_differences[48]; - int32_t previous_reconstructed_sample; - int32_t predicted_difference; - int32_t predicted_sample; -} Prediction; - -typedef struct { - int32_t codeword_history; - int32_t dither_parity; - int32_t dither[NB_SUBBANDS]; - - QMFAnalysis qmf; - Quantize quantize[NB_SUBBANDS]; - InvertQuantize invert_quantize[NB_SUBBANDS]; - Prediction prediction[NB_SUBBANDS]; -} Channel; - -typedef struct { - int hd; - int block_size; - int32_t sync_idx; - Channel channels[NB_CHANNELS]; - AudioFrameQueue afq; -} AptXContext; +#include "aptx.h" static const int32_t quantize_intervals_LF[65] = { @@ -383,17 +309,7 @@ static const int16_t hd_quantize_factor_select_offset_HF[17] = { 70, 90, 115, 147, 192, 264, 398, 521, 521, }; -typedef const struct { - const int32_t *quantize_intervals; - const int32_t *invert_quantize_dither_factors; - const int32_t *quantize_dither_factors; - const int16_t *quantize_factor_select_offset; - int tables_size; - int32_t factor_max; - int32_t prediction_order; -} ConstTables; - -static ConstTables tables[2][NB_SUBBANDS] = { +ConstTables ff_aptx_quant_tables[2][NB_SUBBANDS] = { { [LF] = { quantize_intervals_LF, invert_quantize_dither_factors_LF, @@ -456,24 +372,6 @@ static const int16_t quantization_factors[32] = { }; -/* Rounded right shift with optionnal clipping */ -#define RSHIFT_SIZE(size) \ -av_always_inline \ -static int##size##_t rshift##size(int##size##_t value, int shift) \ -{ \ - int##size##_t rounding = (int##size##_t)1 << (shift - 1); \ - int##size##_t mask = ((int##size##_t)1 << (shift + 1)) - 1; \ - return ((value + rounding) >> shift) - ((value & mask) == rounding); \ -} \ -av_always_inline \ -static int##size##_t rshift##size##_clip24(int##size##_t value, int shift) \ -{ \ - return av_clip_intp2(rshift##size(value, shift), 23); \ -} -RSHIFT_SIZE(32) -RSHIFT_SIZE(64) - - av_always_inline static void aptx_update_codeword_history(Channel *channel) { @@ -483,7 +381,7 @@ static void aptx_update_codeword_history(Channel *channel) channel->codeword_history = (cw << 8) + ((unsigned)channel->codeword_history << 4); } -static void aptx_generate_dither(Channel *channel) +void ff_aptx_generate_dither(Channel *channel) { int subband; int64_t m; @@ -498,256 +396,6 @@ static void aptx_generate_dither(Channel *channel) channel->dither_parity = (d >> 25) & 1; } -/* - * Convolution filter coefficients for the outer QMF of the QMF tree. - * The 2 sets are a mirror of each other. - */ -static const int32_t aptx_qmf_outer_coeffs[NB_FILTERS][FILTER_TAPS] = { - { - 730, -413, -9611, 43626, -121026, 269973, -585547, 2801966, - 697128, -160481, 27611, 8478, -10043, 3511, 688, -897, - }, - { - -897, 688, 3511, -10043, 8478, 27611, -160481, 697128, - 2801966, -585547, 269973, -121026, 43626, -9611, -413, 730, - }, -}; - -/* - * Convolution filter coefficients for the inner QMF of the QMF tree. - * The 2 sets are a mirror of each other. - */ -static const int32_t aptx_qmf_inner_coeffs[NB_FILTERS][FILTER_TAPS] = { - { - 1033, -584, -13592, 61697, -171156, 381799, -828088, 3962579, - 985888, -226954, 39048, 11990, -14203, 4966, 973, -1268, - }, - { - -1268, 973, 4966, -14203, 11990, 39048, -226954, 985888, - 3962579, -828088, 381799, -171156, 61697, -13592, -584, 1033, - }, -}; - -/* - * Push one sample into a circular signal buffer. - */ -av_always_inline -static void aptx_qmf_filter_signal_push(FilterSignal *signal, int32_t sample) -{ - signal->buffer[signal->pos ] = sample; - signal->buffer[signal->pos+FILTER_TAPS] = sample; - signal->pos = (signal->pos + 1) & (FILTER_TAPS - 1); -} - -/* - * Compute the convolution of the signal with the coefficients, and reduce - * to 24 bits by applying the specified right shifting. - */ -av_always_inline -static int32_t aptx_qmf_convolution(FilterSignal *signal, - const int32_t coeffs[FILTER_TAPS], - int shift) -{ - int32_t *sig = &signal->buffer[signal->pos]; - int64_t e = 0; - int i; - - for (i = 0; i < FILTER_TAPS; i++) - e += MUL64(sig[i], coeffs[i]); - - return rshift64_clip24(e, shift); -} - -/* - * Half-band QMF analysis filter realized with a polyphase FIR filter. - * Split into 2 subbands and downsample by 2. - * So for each pair of samples that goes in, one sample goes out, - * split into 2 separate subbands. - */ -av_always_inline -static void aptx_qmf_polyphase_analysis(FilterSignal signal[NB_FILTERS], - const int32_t coeffs[NB_FILTERS][FILTER_TAPS], - int shift, - int32_t samples[NB_FILTERS], - int32_t *low_subband_output, - int32_t *high_subband_output) -{ - int32_t subbands[NB_FILTERS]; - int i; - - for (i = 0; i < NB_FILTERS; i++) { - aptx_qmf_filter_signal_push(&signal[i], samples[NB_FILTERS-1-i]); - subbands[i] = aptx_qmf_convolution(&signal[i], coeffs[i], shift); - } - - *low_subband_output = av_clip_intp2(subbands[0] + subbands[1], 23); - *high_subband_output = av_clip_intp2(subbands[0] - subbands[1], 23); -} - -/* - * Two stage QMF analysis tree. - * Split 4 input samples into 4 subbands and downsample by 4. - * So for each group of 4 samples that goes in, one sample goes out, - * split into 4 separate subbands. - */ -static void aptx_qmf_tree_analysis(QMFAnalysis *qmf, - int32_t samples[4], - int32_t subband_samples[4]) -{ - int32_t intermediate_samples[4]; - int i; - - /* Split 4 input samples into 2 intermediate subbands downsampled to 2 samples */ - for (i = 0; i < 2; i++) - aptx_qmf_polyphase_analysis(qmf->outer_filter_signal, - aptx_qmf_outer_coeffs, 23, - &samples[2*i], - &intermediate_samples[0+i], - &intermediate_samples[2+i]); - - /* Split 2 intermediate subband samples into 4 final subbands downsampled to 1 sample */ - for (i = 0; i < 2; i++) - aptx_qmf_polyphase_analysis(qmf->inner_filter_signal[i], - aptx_qmf_inner_coeffs, 23, - &intermediate_samples[2*i], - &subband_samples[2*i+0], - &subband_samples[2*i+1]); -} - -/* - * Half-band QMF synthesis filter realized with a polyphase FIR filter. - * Join 2 subbands and upsample by 2. - * So for each 2 subbands sample that goes in, a pair of samples goes out. - */ -av_always_inline -static void aptx_qmf_polyphase_synthesis(FilterSignal signal[NB_FILTERS], - const int32_t coeffs[NB_FILTERS][FILTER_TAPS], - int shift, - int32_t low_subband_input, - int32_t high_subband_input, - int32_t samples[NB_FILTERS]) -{ - int32_t subbands[NB_FILTERS]; - int i; - - subbands[0] = low_subband_input + high_subband_input; - subbands[1] = low_subband_input - high_subband_input; - - for (i = 0; i < NB_FILTERS; i++) { - aptx_qmf_filter_signal_push(&signal[i], subbands[1-i]); - samples[i] = aptx_qmf_convolution(&signal[i], coeffs[i], shift); - } -} - -/* - * Two stage QMF synthesis tree. - * Join 4 subbands and upsample by 4. - * So for each 4 subbands sample that goes in, a group of 4 samples goes out. - */ -static void aptx_qmf_tree_synthesis(QMFAnalysis *qmf, - int32_t subband_samples[4], - int32_t samples[4]) -{ - int32_t intermediate_samples[4]; - int i; - - /* Join 4 subbands into 2 intermediate subbands upsampled to 2 samples. */ - for (i = 0; i < 2; i++) - aptx_qmf_polyphase_synthesis(qmf->inner_filter_signal[i], - aptx_qmf_inner_coeffs, 22, - subband_samples[2*i+0], - subband_samples[2*i+1], - &intermediate_samples[2*i]); - - /* Join 2 samples from intermediate subbands upsampled to 4 samples. */ - for (i = 0; i < 2; i++) - aptx_qmf_polyphase_synthesis(qmf->outer_filter_signal, - aptx_qmf_outer_coeffs, 21, - intermediate_samples[0+i], - intermediate_samples[2+i], - &samples[2*i]); -} - - -av_always_inline -static int32_t aptx_bin_search(int32_t value, int32_t factor, - const int32_t *intervals, int32_t nb_intervals) -{ - int32_t idx = 0; - int i; - - for (i = nb_intervals >> 1; i > 0; i >>= 1) - if (MUL64(factor, intervals[idx + i]) <= ((int64_t)value << 24)) - idx += i; - - return idx; -} - -static void aptx_quantize_difference(Quantize *quantize, - int32_t sample_difference, - int32_t dither, - int32_t quantization_factor, - ConstTables *tables) -{ - const int32_t *intervals = tables->quantize_intervals; - int32_t quantized_sample, dithered_sample, parity_change; - int32_t d, mean, interval, inv, sample_difference_abs; - int64_t error; - - sample_difference_abs = FFABS(sample_difference); - sample_difference_abs = FFMIN(sample_difference_abs, (1 << 23) - 1); - - quantized_sample = aptx_bin_search(sample_difference_abs >> 4, - quantization_factor, - intervals, tables->tables_size); - - d = rshift32_clip24(MULH(dither, dither), 7) - (1 << 23); - d = rshift64(MUL64(d, tables->quantize_dither_factors[quantized_sample]), 23); - - intervals += quantized_sample; - mean = (intervals[1] + intervals[0]) / 2; - interval = (intervals[1] - intervals[0]) * (-(sample_difference < 0) | 1); - - dithered_sample = rshift64_clip24(MUL64(dither, interval) + ((int64_t)av_clip_intp2(mean + d, 23) << 32), 32); - error = ((int64_t)sample_difference_abs << 20) - MUL64(dithered_sample, quantization_factor); - quantize->error = FFABS(rshift64(error, 23)); - - parity_change = quantized_sample; - if (error < 0) - quantized_sample--; - else - parity_change--; - - inv = -(sample_difference < 0); - quantize->quantized_sample = quantized_sample ^ inv; - quantize->quantized_sample_parity_change = parity_change ^ inv; -} - -static void aptx_encode_channel(Channel *channel, int32_t samples[4], int hd) -{ - int32_t subband_samples[4]; - int subband; - aptx_qmf_tree_analysis(&channel->qmf, samples, subband_samples); - aptx_generate_dither(channel); - for (subband = 0; subband < NB_SUBBANDS; subband++) { - int32_t diff = av_clip_intp2(subband_samples[subband] - channel->prediction[subband].predicted_sample, 23); - aptx_quantize_difference(&channel->quantize[subband], diff, - channel->dither[subband], - channel->invert_quantize[subband].quantization_factor, - &tables[hd][subband]); - } -} - -static void aptx_decode_channel(Channel *channel, int32_t samples[4]) -{ - int32_t subband_samples[4]; - int subband; - for (subband = 0; subband < NB_SUBBANDS; subband++) - subband_samples[subband] = channel->prediction[subband].previous_reconstructed_sample; - aptx_qmf_tree_synthesis(&channel->qmf, subband_samples, samples); -} - - static void aptx_invert_quantization(InvertQuantize *invert_quantize, int32_t quantized_sample, int32_t dither, ConstTables *tables) @@ -845,7 +493,7 @@ static void aptx_process_subband(InvertQuantize *invert_quantize, tables->prediction_order); } -static void aptx_invert_quantize_and_prediction(Channel *channel, int hd) +void ff_aptx_invert_quantize_and_prediction(Channel *channel, int hd) { int subband; for (subband = 0; subband < NB_SUBBANDS; subband++) @@ -853,138 +501,10 @@ static void aptx_invert_quantize_and_prediction(Channel *channel, int hd) &channel->prediction[subband], channel->quantize[subband].quantized_sample, channel->dither[subband], - &tables[hd][subband]); -} - -static int32_t aptx_quantized_parity(Channel *channel) -{ - int32_t parity = channel->dither_parity; - int subband; - - for (subband = 0; subband < NB_SUBBANDS; subband++) - parity ^= channel->quantize[subband].quantized_sample; - - return parity & 1; + &ff_aptx_quant_tables[hd][subband]); } -/* For each sample, ensure that the parity of all subbands of all channels - * is 0 except once every 8 samples where the parity is forced to 1. */ -static int aptx_check_parity(Channel channels[NB_CHANNELS], int32_t *idx) -{ - int32_t parity = aptx_quantized_parity(&channels[LEFT]) - ^ aptx_quantized_parity(&channels[RIGHT]); - - int eighth = *idx == 7; - *idx = (*idx + 1) & 7; - - return parity ^ eighth; -} - -static void aptx_insert_sync(Channel channels[NB_CHANNELS], int32_t *idx) -{ - if (aptx_check_parity(channels, idx)) { - int i; - Channel *c; - static const int map[] = { 1, 2, 0, 3 }; - Quantize *min = &channels[NB_CHANNELS-1].quantize[map[0]]; - for (c = &channels[NB_CHANNELS-1]; c >= channels; c--) - for (i = 0; i < NB_SUBBANDS; i++) - if (c->quantize[map[i]].error < min->error) - min = &c->quantize[map[i]]; - - /* Forcing the desired parity is done by offsetting by 1 the quantized - * sample from the subband featuring the smallest quantization error. */ - min->quantized_sample = min->quantized_sample_parity_change; - } -} - -static uint16_t aptx_pack_codeword(Channel *channel) -{ - int32_t parity = aptx_quantized_parity(channel); - return (((channel->quantize[3].quantized_sample & 0x06) | parity) << 13) - | (((channel->quantize[2].quantized_sample & 0x03) ) << 11) - | (((channel->quantize[1].quantized_sample & 0x0F) ) << 7) - | (((channel->quantize[0].quantized_sample & 0x7F) ) << 0); -} - -static uint32_t aptxhd_pack_codeword(Channel *channel) -{ - int32_t parity = aptx_quantized_parity(channel); - return (((channel->quantize[3].quantized_sample & 0x01E) | parity) << 19) - | (((channel->quantize[2].quantized_sample & 0x00F) ) << 15) - | (((channel->quantize[1].quantized_sample & 0x03F) ) << 9) - | (((channel->quantize[0].quantized_sample & 0x1FF) ) << 0); -} - -static void aptx_unpack_codeword(Channel *channel, uint16_t codeword) -{ - channel->quantize[0].quantized_sample = sign_extend(codeword >> 0, 7); - channel->quantize[1].quantized_sample = sign_extend(codeword >> 7, 4); - channel->quantize[2].quantized_sample = sign_extend(codeword >> 11, 2); - channel->quantize[3].quantized_sample = sign_extend(codeword >> 13, 3); - channel->quantize[3].quantized_sample = (channel->quantize[3].quantized_sample & ~1) - | aptx_quantized_parity(channel); -} - -static void aptxhd_unpack_codeword(Channel *channel, uint32_t codeword) -{ - channel->quantize[0].quantized_sample = sign_extend(codeword >> 0, 9); - channel->quantize[1].quantized_sample = sign_extend(codeword >> 9, 6); - channel->quantize[2].quantized_sample = sign_extend(codeword >> 15, 4); - channel->quantize[3].quantized_sample = sign_extend(codeword >> 19, 5); - channel->quantize[3].quantized_sample = (channel->quantize[3].quantized_sample & ~1) - | aptx_quantized_parity(channel); -} - -static void aptx_encode_samples(AptXContext *ctx, - int32_t samples[NB_CHANNELS][4], - uint8_t *output) -{ - int channel; - for (channel = 0; channel < NB_CHANNELS; channel++) - aptx_encode_channel(&ctx->channels[channel], samples[channel], ctx->hd); - - aptx_insert_sync(ctx->channels, &ctx->sync_idx); - - for (channel = 0; channel < NB_CHANNELS; channel++) { - aptx_invert_quantize_and_prediction(&ctx->channels[channel], ctx->hd); - if (ctx->hd) - AV_WB24(output + 3*channel, - aptxhd_pack_codeword(&ctx->channels[channel])); - else - AV_WB16(output + 2*channel, - aptx_pack_codeword(&ctx->channels[channel])); - } -} - -static int aptx_decode_samples(AptXContext *ctx, - const uint8_t *input, - int32_t samples[NB_CHANNELS][4]) -{ - int channel, ret; - - for (channel = 0; channel < NB_CHANNELS; channel++) { - aptx_generate_dither(&ctx->channels[channel]); - - if (ctx->hd) - aptxhd_unpack_codeword(&ctx->channels[channel], - AV_RB24(input + 3*channel)); - else - aptx_unpack_codeword(&ctx->channels[channel], - AV_RB16(input + 2*channel)); - aptx_invert_quantize_and_prediction(&ctx->channels[channel], ctx->hd); - } - - ret = aptx_check_parity(ctx->channels, &ctx->sync_idx); - - for (channel = 0; channel < NB_CHANNELS; channel++) - aptx_decode_channel(&ctx->channels[channel], samples[channel]); - - return ret; -} - - -static av_cold int aptx_init(AVCodecContext *avctx) +av_cold int ff_aptx_init(AVCodecContext *avctx) { AptXContext *s = avctx->priv_data; int chan, subband; @@ -1017,149 +537,9 @@ static av_cold int aptx_init(AVCodecContext *avctx) return 0; } -static int aptx_decode_frame(AVCodecContext *avctx, void *data, - int *got_frame_ptr, AVPacket *avpkt) -{ - AptXContext *s = avctx->priv_data; - AVFrame *frame = data; - int pos, opos, channel, sample, ret; - - if (avpkt->size < s->block_size) { - av_log(avctx, AV_LOG_ERROR, "Packet is too small\n"); - return AVERROR_INVALIDDATA; - } - - /* get output buffer */ - frame->channels = NB_CHANNELS; - frame->format = AV_SAMPLE_FMT_S32P; - frame->nb_samples = 4 * avpkt->size / s->block_size; - if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) - return ret; - - for (pos = 0, opos = 0; opos < frame->nb_samples; pos += s->block_size, opos += 4) { - int32_t samples[NB_CHANNELS][4]; - - if (aptx_decode_samples(s, &avpkt->data[pos], samples)) { - av_log(avctx, AV_LOG_ERROR, "Synchronization error\n"); - return AVERROR_INVALIDDATA; - } - - for (channel = 0; channel < NB_CHANNELS; channel++) - for (sample = 0; sample < 4; sample++) - AV_WN32A(&frame->data[channel][4*(opos+sample)], - samples[channel][sample] * 256); - } - - *got_frame_ptr = 1; - return s->block_size * frame->nb_samples / 4; -} - -static int aptx_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, - const AVFrame *frame, int *got_packet_ptr) -{ - AptXContext *s = avctx->priv_data; - int pos, ipos, channel, sample, output_size, ret; - - if ((ret = ff_af_queue_add(&s->afq, frame)) < 0) - return ret; - - output_size = s->block_size * frame->nb_samples/4; - if ((ret = ff_alloc_packet2(avctx, avpkt, output_size, 0)) < 0) - return ret; - - for (pos = 0, ipos = 0; pos < output_size; pos += s->block_size, ipos += 4) { - int32_t samples[NB_CHANNELS][4]; - - for (channel = 0; channel < NB_CHANNELS; channel++) - for (sample = 0; sample < 4; sample++) - samples[channel][sample] = (int32_t)AV_RN32A(&frame->data[channel][4*(ipos+sample)]) >> 8; - - aptx_encode_samples(s, samples, avpkt->data + pos); - } - - ff_af_queue_remove(&s->afq, frame->nb_samples, &avpkt->pts, &avpkt->duration); - *got_packet_ptr = 1; - return 0; -} - -static av_cold int aptx_close(AVCodecContext *avctx) +av_cold int ff_aptx_close(AVCodecContext *avctx) { AptXContext *s = avctx->priv_data; ff_af_queue_close(&s->afq); return 0; } - - -#if CONFIG_APTX_DECODER -AVCodec ff_aptx_decoder = { - .name = "aptx", - .long_name = NULL_IF_CONFIG_SMALL("aptX (Audio Processing Technology for Bluetooth)"), - .type = AVMEDIA_TYPE_AUDIO, - .id = AV_CODEC_ID_APTX, - .priv_data_size = sizeof(AptXContext), - .init = aptx_init, - .decode = aptx_decode_frame, - .close = aptx_close, - .capabilities = AV_CODEC_CAP_DR1, - .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, - .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_STEREO, 0}, - .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P, - AV_SAMPLE_FMT_NONE }, -}; -#endif - -#if CONFIG_APTX_HD_DECODER -AVCodec ff_aptx_hd_decoder = { - .name = "aptx_hd", - .long_name = NULL_IF_CONFIG_SMALL("aptX HD (Audio Processing Technology for Bluetooth)"), - .type = AVMEDIA_TYPE_AUDIO, - .id = AV_CODEC_ID_APTX_HD, - .priv_data_size = sizeof(AptXContext), - .init = aptx_init, - .decode = aptx_decode_frame, - .close = aptx_close, - .capabilities = AV_CODEC_CAP_DR1, - .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, - .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_STEREO, 0}, - .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P, - AV_SAMPLE_FMT_NONE }, -}; -#endif - -#if CONFIG_APTX_ENCODER -AVCodec ff_aptx_encoder = { - .name = "aptx", - .long_name = NULL_IF_CONFIG_SMALL("aptX (Audio Processing Technology for Bluetooth)"), - .type = AVMEDIA_TYPE_AUDIO, - .id = AV_CODEC_ID_APTX, - .priv_data_size = sizeof(AptXContext), - .init = aptx_init, - .encode2 = aptx_encode_frame, - .close = aptx_close, - .capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME, - .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, - .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_STEREO, 0}, - .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P, - AV_SAMPLE_FMT_NONE }, - .supported_samplerates = (const int[]) {8000, 16000, 24000, 32000, 44100, 48000, 0}, -}; -#endif - -#if CONFIG_APTX_HD_ENCODER -AVCodec ff_aptx_hd_encoder = { - .name = "aptx_hd", - .long_name = NULL_IF_CONFIG_SMALL("aptX HD (Audio Processing Technology for Bluetooth)"), - .type = AVMEDIA_TYPE_AUDIO, - .id = AV_CODEC_ID_APTX_HD, - .priv_data_size = sizeof(AptXContext), - .init = aptx_init, - .encode2 = aptx_encode_frame, - .close = aptx_close, - .capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME, - .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, - .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_STEREO, 0}, - .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P, - AV_SAMPLE_FMT_NONE }, - .supported_samplerates = (const int[]) {8000, 16000, 24000, 32000, 44100, 48000, 0}, -}; -#endif diff --git a/libavcodec/aptx.h b/libavcodec/aptx.h new file mode 100644 index 0000000000..2ec18d37ec --- /dev/null +++ b/libavcodec/aptx.h @@ -0,0 +1,221 @@ +/* + * Audio Processing Technology codec for Bluetooth (aptX) + * + * Copyright (C) 2017 Aurelien Jacobs + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVCODEC_APTX_H +#define AVCODEC_APTX_H + +#include "libavutil/intreadwrite.h" +#include "avcodec.h" +#include "internal.h" +#include "mathops.h" +#include "audio_frame_queue.h" + + +enum channels { + LEFT, + RIGHT, + NB_CHANNELS +}; + +enum subbands { + LF, // Low Frequency (0-5.5 kHz) + MLF, // Medium-Low Frequency (5.5-11kHz) + MHF, // Medium-High Frequency (11-16.5kHz) + HF, // High Frequency (16.5-22kHz) + NB_SUBBANDS +}; + +#define NB_FILTERS 2 +#define FILTER_TAPS 16 + +typedef struct { + int pos; + int32_t buffer[2*FILTER_TAPS]; +} FilterSignal; + +typedef struct { + FilterSignal outer_filter_signal[NB_FILTERS]; + FilterSignal inner_filter_signal[NB_FILTERS][NB_FILTERS]; +} QMFAnalysis; + +typedef struct { + int32_t quantized_sample; + int32_t quantized_sample_parity_change; + int32_t error; +} Quantize; + +typedef struct { + int32_t quantization_factor; + int32_t factor_select; + int32_t reconstructed_difference; +} InvertQuantize; + +typedef struct { + int32_t prev_sign[2]; + int32_t s_weight[2]; + int32_t d_weight[24]; + int32_t pos; + int32_t reconstructed_differences[48]; + int32_t previous_reconstructed_sample; + int32_t predicted_difference; + int32_t predicted_sample; +} Prediction; + +typedef struct { + int32_t codeword_history; + int32_t dither_parity; + int32_t dither[NB_SUBBANDS]; + + QMFAnalysis qmf; + Quantize quantize[NB_SUBBANDS]; + InvertQuantize invert_quantize[NB_SUBBANDS]; + Prediction prediction[NB_SUBBANDS]; +} Channel; + +typedef struct { + int hd; + int block_size; + int32_t sync_idx; + Channel channels[NB_CHANNELS]; + AudioFrameQueue afq; +} AptXContext; + +typedef const struct { + const int32_t *quantize_intervals; + const int32_t *invert_quantize_dither_factors; + const int32_t *quantize_dither_factors; + const int16_t *quantize_factor_select_offset; + int tables_size; + int32_t factor_max; + int32_t prediction_order; +} ConstTables; + +extern ConstTables ff_aptx_quant_tables[2][NB_SUBBANDS]; + +/* Rounded right shift with optionnal clipping */ +#define RSHIFT_SIZE(size) \ +av_always_inline \ +static int##size##_t rshift##size(int##size##_t value, int shift) \ +{ \ + int##size##_t rounding = (int##size##_t)1 << (shift - 1); \ + int##size##_t mask = ((int##size##_t)1 << (shift + 1)) - 1; \ + return ((value + rounding) >> shift) - ((value & mask) == rounding); \ +} \ +av_always_inline \ +static int##size##_t rshift##size##_clip24(int##size##_t value, int shift) \ +{ \ + return av_clip_intp2(rshift##size(value, shift), 23); \ +} +RSHIFT_SIZE(32) +RSHIFT_SIZE(64) + +/* + * Convolution filter coefficients for the outer QMF of the QMF tree. + * The 2 sets are a mirror of each other. + */ +static const int32_t aptx_qmf_outer_coeffs[NB_FILTERS][FILTER_TAPS] = { + { + 730, -413, -9611, 43626, -121026, 269973, -585547, 2801966, + 697128, -160481, 27611, 8478, -10043, 3511, 688, -897, + }, + { + -897, 688, 3511, -10043, 8478, 27611, -160481, 697128, + 2801966, -585547, 269973, -121026, 43626, -9611, -413, 730, + }, +}; + +/* + * Convolution filter coefficients for the inner QMF of the QMF tree. + * The 2 sets are a mirror of each other. + */ +static const int32_t aptx_qmf_inner_coeffs[NB_FILTERS][FILTER_TAPS] = { + { + 1033, -584, -13592, 61697, -171156, 381799, -828088, 3962579, + 985888, -226954, 39048, 11990, -14203, 4966, 973, -1268, + }, + { + -1268, 973, 4966, -14203, 11990, 39048, -226954, 985888, + 3962579, -828088, 381799, -171156, 61697, -13592, -584, 1033, + }, +}; + +/* + * Push one sample into a circular signal buffer. + */ +av_always_inline +static void aptx_qmf_filter_signal_push(FilterSignal *signal, int32_t sample) +{ + signal->buffer[signal->pos ] = sample; + signal->buffer[signal->pos+FILTER_TAPS] = sample; + signal->pos = (signal->pos + 1) & (FILTER_TAPS - 1); +} + +/* + * Compute the convolution of the signal with the coefficients, and reduce + * to 24 bits by applying the specified right shifting. + */ +av_always_inline +static int32_t aptx_qmf_convolution(FilterSignal *signal, + const int32_t coeffs[FILTER_TAPS], + int shift) +{ + int32_t *sig = &signal->buffer[signal->pos]; + int64_t e = 0; + int i; + + for (i = 0; i < FILTER_TAPS; i++) + e += MUL64(sig[i], coeffs[i]); + + return rshift64_clip24(e, shift); +} + +static inline int32_t aptx_quantized_parity(Channel *channel) +{ + int32_t parity = channel->dither_parity; + int subband; + + for (subband = 0; subband < NB_SUBBANDS; subband++) + parity ^= channel->quantize[subband].quantized_sample; + + return parity & 1; +} + +/* For each sample, ensure that the parity of all subbands of all channels + * is 0 except once every 8 samples where the parity is forced to 1. */ +static inline int aptx_check_parity(Channel channels[NB_CHANNELS], int32_t *idx) +{ + int32_t parity = aptx_quantized_parity(&channels[LEFT]) + ^ aptx_quantized_parity(&channels[RIGHT]); + + int eighth = *idx == 7; + *idx = (*idx + 1) & 7; + + return parity ^ eighth; +} + +void ff_aptx_invert_quantize_and_prediction(Channel *channel, int hd); +void ff_aptx_generate_dither(Channel *channel); + +int ff_aptx_init(AVCodecContext *avctx); +int ff_aptx_close(AVCodecContext *avctx); + +#endif /* AVCODEC_APTX_H */ diff --git a/libavcodec/aptxdec.c b/libavcodec/aptxdec.c new file mode 100644 index 0000000000..6aaa6f00bd --- /dev/null +++ b/libavcodec/aptxdec.c @@ -0,0 +1,206 @@ +/* + * Audio Processing Technology codec for Bluetooth (aptX) + * + * Copyright (C) 2017 Aurelien Jacobs + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "aptx.h" + +/* + * Half-band QMF synthesis filter realized with a polyphase FIR filter. + * Join 2 subbands and upsample by 2. + * So for each 2 subbands sample that goes in, a pair of samples goes out. + */ +av_always_inline +static void aptx_qmf_polyphase_synthesis(FilterSignal signal[NB_FILTERS], + const int32_t coeffs[NB_FILTERS][FILTER_TAPS], + int shift, + int32_t low_subband_input, + int32_t high_subband_input, + int32_t samples[NB_FILTERS]) +{ + int32_t subbands[NB_FILTERS]; + int i; + + subbands[0] = low_subband_input + high_subband_input; + subbands[1] = low_subband_input - high_subband_input; + + for (i = 0; i < NB_FILTERS; i++) { + aptx_qmf_filter_signal_push(&signal[i], subbands[1-i]); + samples[i] = aptx_qmf_convolution(&signal[i], coeffs[i], shift); + } +} + +/* + * Two stage QMF synthesis tree. + * Join 4 subbands and upsample by 4. + * So for each 4 subbands sample that goes in, a group of 4 samples goes out. + */ +static void aptx_qmf_tree_synthesis(QMFAnalysis *qmf, + int32_t subband_samples[4], + int32_t samples[4]) +{ + int32_t intermediate_samples[4]; + int i; + + /* Join 4 subbands into 2 intermediate subbands upsampled to 2 samples. */ + for (i = 0; i < 2; i++) + aptx_qmf_polyphase_synthesis(qmf->inner_filter_signal[i], + aptx_qmf_inner_coeffs, 22, + subband_samples[2*i+0], + subband_samples[2*i+1], + &intermediate_samples[2*i]); + + /* Join 2 samples from intermediate subbands upsampled to 4 samples. */ + for (i = 0; i < 2; i++) + aptx_qmf_polyphase_synthesis(qmf->outer_filter_signal, + aptx_qmf_outer_coeffs, 21, + intermediate_samples[0+i], + intermediate_samples[2+i], + &samples[2*i]); +} + + +static void aptx_decode_channel(Channel *channel, int32_t samples[4]) +{ + int32_t subband_samples[4]; + int subband; + for (subband = 0; subband < NB_SUBBANDS; subband++) + subband_samples[subband] = channel->prediction[subband].previous_reconstructed_sample; + aptx_qmf_tree_synthesis(&channel->qmf, subband_samples, samples); +} + +static void aptx_unpack_codeword(Channel *channel, uint16_t codeword) +{ + channel->quantize[0].quantized_sample = sign_extend(codeword >> 0, 7); + channel->quantize[1].quantized_sample = sign_extend(codeword >> 7, 4); + channel->quantize[2].quantized_sample = sign_extend(codeword >> 11, 2); + channel->quantize[3].quantized_sample = sign_extend(codeword >> 13, 3); + channel->quantize[3].quantized_sample = (channel->quantize[3].quantized_sample & ~1) + | aptx_quantized_parity(channel); +} + +static void aptxhd_unpack_codeword(Channel *channel, uint32_t codeword) +{ + channel->quantize[0].quantized_sample = sign_extend(codeword >> 0, 9); + channel->quantize[1].quantized_sample = sign_extend(codeword >> 9, 6); + channel->quantize[2].quantized_sample = sign_extend(codeword >> 15, 4); + channel->quantize[3].quantized_sample = sign_extend(codeword >> 19, 5); + channel->quantize[3].quantized_sample = (channel->quantize[3].quantized_sample & ~1) + | aptx_quantized_parity(channel); +} + +static int aptx_decode_samples(AptXContext *ctx, + const uint8_t *input, + int32_t samples[NB_CHANNELS][4]) +{ + int channel, ret; + + for (channel = 0; channel < NB_CHANNELS; channel++) { + ff_aptx_generate_dither(&ctx->channels[channel]); + + if (ctx->hd) + aptxhd_unpack_codeword(&ctx->channels[channel], + AV_RB24(input + 3*channel)); + else + aptx_unpack_codeword(&ctx->channels[channel], + AV_RB16(input + 2*channel)); + ff_aptx_invert_quantize_and_prediction(&ctx->channels[channel], ctx->hd); + } + + ret = aptx_check_parity(ctx->channels, &ctx->sync_idx); + + for (channel = 0; channel < NB_CHANNELS; channel++) + aptx_decode_channel(&ctx->channels[channel], samples[channel]); + + return ret; +} + +static int aptx_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) +{ + AptXContext *s = avctx->priv_data; + AVFrame *frame = data; + int pos, opos, channel, sample, ret; + + if (avpkt->size < s->block_size) { + av_log(avctx, AV_LOG_ERROR, "Packet is too small\n"); + return AVERROR_INVALIDDATA; + } + + /* get output buffer */ + frame->channels = NB_CHANNELS; + frame->format = AV_SAMPLE_FMT_S32P; + frame->nb_samples = 4 * avpkt->size / s->block_size; + if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) + return ret; + + for (pos = 0, opos = 0; opos < frame->nb_samples; pos += s->block_size, opos += 4) { + int32_t samples[NB_CHANNELS][4]; + + if (aptx_decode_samples(s, &avpkt->data[pos], samples)) { + av_log(avctx, AV_LOG_ERROR, "Synchronization error\n"); + return AVERROR_INVALIDDATA; + } + + for (channel = 0; channel < NB_CHANNELS; channel++) + for (sample = 0; sample < 4; sample++) + AV_WN32A(&frame->data[channel][4*(opos+sample)], + samples[channel][sample] * 256); + } + + *got_frame_ptr = 1; + return s->block_size * frame->nb_samples / 4; +} + +#if CONFIG_APTX_DECODER +AVCodec ff_aptx_decoder = { + .name = "aptx", + .long_name = NULL_IF_CONFIG_SMALL("aptX (Audio Processing Technology for Bluetooth)"), + .type = AVMEDIA_TYPE_AUDIO, + .id = AV_CODEC_ID_APTX, + .priv_data_size = sizeof(AptXContext), + .init = ff_aptx_init, + .decode = aptx_decode_frame, + .close = ff_aptx_close, + .capabilities = AV_CODEC_CAP_DR1, + .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, + .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_STEREO, 0}, + .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P, + AV_SAMPLE_FMT_NONE }, +}; +#endif + +#if CONFIG_APTX_HD_DECODER +AVCodec ff_aptx_hd_decoder = { + .name = "aptx_hd", + .long_name = NULL_IF_CONFIG_SMALL("aptX HD (Audio Processing Technology for Bluetooth)"), + .type = AVMEDIA_TYPE_AUDIO, + .id = AV_CODEC_ID_APTX_HD, + .priv_data_size = sizeof(AptXContext), + .init = ff_aptx_init, + .decode = aptx_decode_frame, + .close = ff_aptx_close, + .capabilities = AV_CODEC_CAP_DR1, + .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, + .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_STEREO, 0}, + .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P, + AV_SAMPLE_FMT_NONE }, +}; +#endif diff --git a/libavcodec/aptxenc.c b/libavcodec/aptxenc.c new file mode 100644 index 0000000000..6e2ceef40d --- /dev/null +++ b/libavcodec/aptxenc.c @@ -0,0 +1,271 @@ +/* + * Audio Processing Technology codec for Bluetooth (aptX) + * + * Copyright (C) 2017 Aurelien Jacobs + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "aptx.h" + +/* + * Half-band QMF analysis filter realized with a polyphase FIR filter. + * Split into 2 subbands and downsample by 2. + * So for each pair of samples that goes in, one sample goes out, + * split into 2 separate subbands. + */ +av_always_inline +static void aptx_qmf_polyphase_analysis(FilterSignal signal[NB_FILTERS], + const int32_t coeffs[NB_FILTERS][FILTER_TAPS], + int shift, + int32_t samples[NB_FILTERS], + int32_t *low_subband_output, + int32_t *high_subband_output) +{ + int32_t subbands[NB_FILTERS]; + int i; + + for (i = 0; i < NB_FILTERS; i++) { + aptx_qmf_filter_signal_push(&signal[i], samples[NB_FILTERS-1-i]); + subbands[i] = aptx_qmf_convolution(&signal[i], coeffs[i], shift); + } + + *low_subband_output = av_clip_intp2(subbands[0] + subbands[1], 23); + *high_subband_output = av_clip_intp2(subbands[0] - subbands[1], 23); +} + +/* + * Two stage QMF analysis tree. + * Split 4 input samples into 4 subbands and downsample by 4. + * So for each group of 4 samples that goes in, one sample goes out, + * split into 4 separate subbands. + */ +static void aptx_qmf_tree_analysis(QMFAnalysis *qmf, + int32_t samples[4], + int32_t subband_samples[4]) +{ + int32_t intermediate_samples[4]; + int i; + + /* Split 4 input samples into 2 intermediate subbands downsampled to 2 samples */ + for (i = 0; i < 2; i++) + aptx_qmf_polyphase_analysis(qmf->outer_filter_signal, + aptx_qmf_outer_coeffs, 23, + &samples[2*i], + &intermediate_samples[0+i], + &intermediate_samples[2+i]); + + /* Split 2 intermediate subband samples into 4 final subbands downsampled to 1 sample */ + for (i = 0; i < 2; i++) + aptx_qmf_polyphase_analysis(qmf->inner_filter_signal[i], + aptx_qmf_inner_coeffs, 23, + &intermediate_samples[2*i], + &subband_samples[2*i+0], + &subband_samples[2*i+1]); +} + +av_always_inline +static int32_t aptx_bin_search(int32_t value, int32_t factor, + const int32_t *intervals, int32_t nb_intervals) +{ + int32_t idx = 0; + int i; + + for (i = nb_intervals >> 1; i > 0; i >>= 1) + if (MUL64(factor, intervals[idx + i]) <= ((int64_t)value << 24)) + idx += i; + + return idx; +} + +static void aptx_quantize_difference(Quantize *quantize, + int32_t sample_difference, + int32_t dither, + int32_t quantization_factor, + ConstTables *tables) +{ + const int32_t *intervals = tables->quantize_intervals; + int32_t quantized_sample, dithered_sample, parity_change; + int32_t d, mean, interval, inv, sample_difference_abs; + int64_t error; + + sample_difference_abs = FFABS(sample_difference); + sample_difference_abs = FFMIN(sample_difference_abs, (1 << 23) - 1); + + quantized_sample = aptx_bin_search(sample_difference_abs >> 4, + quantization_factor, + intervals, tables->tables_size); + + d = rshift32_clip24(MULH(dither, dither), 7) - (1 << 23); + d = rshift64(MUL64(d, tables->quantize_dither_factors[quantized_sample]), 23); + + intervals += quantized_sample; + mean = (intervals[1] + intervals[0]) / 2; + interval = (intervals[1] - intervals[0]) * (-(sample_difference < 0) | 1); + + dithered_sample = rshift64_clip24(MUL64(dither, interval) + ((int64_t)av_clip_intp2(mean + d, 23) << 32), 32); + error = ((int64_t)sample_difference_abs << 20) - MUL64(dithered_sample, quantization_factor); + quantize->error = FFABS(rshift64(error, 23)); + + parity_change = quantized_sample; + if (error < 0) + quantized_sample--; + else + parity_change--; + + inv = -(sample_difference < 0); + quantize->quantized_sample = quantized_sample ^ inv; + quantize->quantized_sample_parity_change = parity_change ^ inv; +} + +static void aptx_encode_channel(Channel *channel, int32_t samples[4], int hd) +{ + int32_t subband_samples[4]; + int subband; + aptx_qmf_tree_analysis(&channel->qmf, samples, subband_samples); + ff_aptx_generate_dither(channel); + for (subband = 0; subband < NB_SUBBANDS; subband++) { + int32_t diff = av_clip_intp2(subband_samples[subband] - channel->prediction[subband].predicted_sample, 23); + aptx_quantize_difference(&channel->quantize[subband], diff, + channel->dither[subband], + channel->invert_quantize[subband].quantization_factor, + &ff_aptx_quant_tables[hd][subband]); + } +} + +static void aptx_insert_sync(Channel channels[NB_CHANNELS], int32_t *idx) +{ + if (aptx_check_parity(channels, idx)) { + int i; + Channel *c; + static const int map[] = { 1, 2, 0, 3 }; + Quantize *min = &channels[NB_CHANNELS-1].quantize[map[0]]; + for (c = &channels[NB_CHANNELS-1]; c >= channels; c--) + for (i = 0; i < NB_SUBBANDS; i++) + if (c->quantize[map[i]].error < min->error) + min = &c->quantize[map[i]]; + + /* Forcing the desired parity is done by offsetting by 1 the quantized + * sample from the subband featuring the smallest quantization error. */ + min->quantized_sample = min->quantized_sample_parity_change; + } +} + +static uint16_t aptx_pack_codeword(Channel *channel) +{ + int32_t parity = aptx_quantized_parity(channel); + return (((channel->quantize[3].quantized_sample & 0x06) | parity) << 13) + | (((channel->quantize[2].quantized_sample & 0x03) ) << 11) + | (((channel->quantize[1].quantized_sample & 0x0F) ) << 7) + | (((channel->quantize[0].quantized_sample & 0x7F) ) << 0); +} + +static uint32_t aptxhd_pack_codeword(Channel *channel) +{ + int32_t parity = aptx_quantized_parity(channel); + return (((channel->quantize[3].quantized_sample & 0x01E) | parity) << 19) + | (((channel->quantize[2].quantized_sample & 0x00F) ) << 15) + | (((channel->quantize[1].quantized_sample & 0x03F) ) << 9) + | (((channel->quantize[0].quantized_sample & 0x1FF) ) << 0); +} + +static void aptx_encode_samples(AptXContext *ctx, + int32_t samples[NB_CHANNELS][4], + uint8_t *output) +{ + int channel; + for (channel = 0; channel < NB_CHANNELS; channel++) + aptx_encode_channel(&ctx->channels[channel], samples[channel], ctx->hd); + + aptx_insert_sync(ctx->channels, &ctx->sync_idx); + + for (channel = 0; channel < NB_CHANNELS; channel++) { + ff_aptx_invert_quantize_and_prediction(&ctx->channels[channel], ctx->hd); + if (ctx->hd) + AV_WB24(output + 3*channel, + aptxhd_pack_codeword(&ctx->channels[channel])); + else + AV_WB16(output + 2*channel, + aptx_pack_codeword(&ctx->channels[channel])); + } +} + +static int aptx_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, + const AVFrame *frame, int *got_packet_ptr) +{ + AptXContext *s = avctx->priv_data; + int pos, ipos, channel, sample, output_size, ret; + + if ((ret = ff_af_queue_add(&s->afq, frame)) < 0) + return ret; + + output_size = s->block_size * frame->nb_samples/4; + if ((ret = ff_alloc_packet2(avctx, avpkt, output_size, 0)) < 0) + return ret; + + for (pos = 0, ipos = 0; pos < output_size; pos += s->block_size, ipos += 4) { + int32_t samples[NB_CHANNELS][4]; + + for (channel = 0; channel < NB_CHANNELS; channel++) + for (sample = 0; sample < 4; sample++) + samples[channel][sample] = (int32_t)AV_RN32A(&frame->data[channel][4*(ipos+sample)]) >> 8; + + aptx_encode_samples(s, samples, avpkt->data + pos); + } + + ff_af_queue_remove(&s->afq, frame->nb_samples, &avpkt->pts, &avpkt->duration); + *got_packet_ptr = 1; + return 0; +} + +#if CONFIG_APTX_ENCODER +AVCodec ff_aptx_encoder = { + .name = "aptx", + .long_name = NULL_IF_CONFIG_SMALL("aptX (Audio Processing Technology for Bluetooth)"), + .type = AVMEDIA_TYPE_AUDIO, + .id = AV_CODEC_ID_APTX, + .priv_data_size = sizeof(AptXContext), + .init = ff_aptx_init, + .encode2 = aptx_encode_frame, + .close = ff_aptx_close, + .capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME, + .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, + .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_STEREO, 0}, + .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P, + AV_SAMPLE_FMT_NONE }, + .supported_samplerates = (const int[]) {8000, 16000, 24000, 32000, 44100, 48000, 0}, +}; +#endif + +#if CONFIG_APTX_HD_ENCODER +AVCodec ff_aptx_hd_encoder = { + .name = "aptx_hd", + .long_name = NULL_IF_CONFIG_SMALL("aptX HD (Audio Processing Technology for Bluetooth)"), + .type = AVMEDIA_TYPE_AUDIO, + .id = AV_CODEC_ID_APTX_HD, + .priv_data_size = sizeof(AptXContext), + .init = ff_aptx_init, + .encode2 = aptx_encode_frame, + .close = ff_aptx_close, + .capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME, + .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, + .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_STEREO, 0}, + .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P, + AV_SAMPLE_FMT_NONE }, + .supported_samplerates = (const int[]) {8000, 16000, 24000, 32000, 44100, 48000, 0}, +}; +#endif