X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Faacenc.c;h=e9f6e2ffbfeb93b6d76437e1930943dc6e505a2b;hb=9a07c1332cfe092b57b5758f22b686ca58806c60;hp=6d66c66862ff329959c08ef080a6723cea38707e;hpb=072c0d605fd4815441dc2f8f4c5f3b4efc81c878;p=ffmpeg diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c index 6d66c66862f..e9f6e2ffbfe 100644 --- a/libavcodec/aacenc.c +++ b/libavcodec/aacenc.c @@ -2,20 +2,20 @@ * AAC encoder * Copyright (C) 2008 Konstantin Shishkov * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav 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, + * Libav 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 + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -30,10 +30,15 @@ * add temporal noise shaping ***********************************/ +#include "libavutil/float_dsp.h" +#include "libavutil/opt.h" #include "avcodec.h" #include "put_bits.h" #include "dsputil.h" +#include "internal.h" #include "mpeg4audio.h" +#include "kbdwin.h" +#include "sinewin.h" #include "aac.h" #include "aactab.h" @@ -41,6 +46,16 @@ #include "psymodel.h" +#define AAC_MAX_CHANNELS 6 + +#define ERROR_IF(cond, ...) \ + if (cond) { \ + av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \ + return AVERROR(EINVAL); \ + } + +float ff_aac_pow34sf_tab[428]; + static const uint8_t swb_size_1024_96[] = { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44, @@ -130,6 +145,18 @@ static const uint8_t aac_chan_configs[6][5] = { {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE }; +/** + * Table to remap channels from Libav's default order to AAC order. + */ +static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = { + { 0 }, + { 0, 1 }, + { 2, 0, 1 }, + { 2, 0, 1, 3 }, + { 2, 0, 1, 3, 4 }, + { 2, 0, 1, 4, 5, 3 }, +}; + /** * Make AAC audio config object. * @see 1.6.2.1 "Syntax - AudioSpecificConfig" @@ -142,123 +169,98 @@ static void put_audio_specific_config(AVCodecContext *avctx) init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8); put_bits(&pb, 5, 2); //object type - AAC-LC put_bits(&pb, 4, s->samplerate_index); //sample rate index - put_bits(&pb, 4, avctx->channels); + put_bits(&pb, 4, s->channels); //GASpecificConfig put_bits(&pb, 1, 0); //frame length - 1024 samples put_bits(&pb, 1, 0); //does not depend on core coder put_bits(&pb, 1, 0); //is not extension + + //Explicitly Mark SBR absent + put_bits(&pb, 11, 0x2b7); //sync extension + put_bits(&pb, 5, AOT_SBR); + put_bits(&pb, 1, 0); flush_put_bits(&pb); } -static av_cold int aac_encode_init(AVCodecContext *avctx) -{ - AACEncContext *s = avctx->priv_data; - int i; - const uint8_t *sizes[2]; - int lengths[2]; +#define WINDOW_FUNC(type) \ +static void apply_ ##type ##_window(DSPContext *dsp, AVFloatDSPContext *fdsp, \ + SingleChannelElement *sce, \ + const float *audio) - avctx->frame_size = 1024; +WINDOW_FUNC(only_long) +{ + const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024; + const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024; + float *out = sce->ret; - for (i = 0; i < 16; i++) - if (avctx->sample_rate == ff_mpeg4audio_sample_rates[i]) - break; - if (i == 16) { - av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate); - return -1; - } - if (avctx->channels > 6) { - av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels); - return -1; - } - if (avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW) { - av_log(avctx, AV_LOG_ERROR, "Unsupported profile %d\n", avctx->profile); - return -1; - } - if (1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * avctx->channels) { - av_log(avctx, AV_LOG_ERROR, "Too many bits per frame requested\n"); - return -1; - } - s->samplerate_index = i; + fdsp->vector_fmul (out, audio, lwindow, 1024); + dsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024); +} - dsputil_init(&s->dsp, avctx); - ff_mdct_init(&s->mdct1024, 11, 0, 1.0); - ff_mdct_init(&s->mdct128, 8, 0, 1.0); - // window init - ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024); - ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128); - ff_init_ff_sine_windows(10); - ff_init_ff_sine_windows(7); +WINDOW_FUNC(long_start) +{ + const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024; + const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; + float *out = sce->ret; + + fdsp->vector_fmul(out, audio, lwindow, 1024); + memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448); + dsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128); + memset(out + 1024 + 576, 0, sizeof(out[0]) * 448); +} - s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0])); - s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]); - avctx->extradata = av_mallocz(2 + FF_INPUT_BUFFER_PADDING_SIZE); - avctx->extradata_size = 2; - put_audio_specific_config(avctx); +WINDOW_FUNC(long_stop) +{ + const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024; + const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; + float *out = sce->ret; + + memset(out, 0, sizeof(out[0]) * 448); + fdsp->vector_fmul(out + 448, audio + 448, swindow, 128); + memcpy(out + 576, audio + 576, sizeof(out[0]) * 448); + dsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024); +} - sizes[0] = swb_size_1024[i]; - sizes[1] = swb_size_128[i]; - lengths[0] = ff_aac_num_swb_1024[i]; - lengths[1] = ff_aac_num_swb_128[i]; - ff_psy_init(&s->psy, avctx, 2, sizes, lengths); - s->psypp = ff_psy_preprocess_init(avctx); - s->coder = &ff_aac_coders[2]; +WINDOW_FUNC(eight_short) +{ + const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; + const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; + const float *in = audio + 448; + float *out = sce->ret; + int w; - s->lambda = avctx->global_quality ? avctx->global_quality : 120; + for (w = 0; w < 8; w++) { + fdsp->vector_fmul (out, in, w ? pwindow : swindow, 128); + out += 128; + in += 128; + dsp->vector_fmul_reverse(out, in, swindow, 128); + out += 128; + } +} - ff_aac_tableinit(); +static void (*const apply_window[4])(DSPContext *dsp, AVFloatDSPContext *fdsp, + SingleChannelElement *sce, + const float *audio) = { + [ONLY_LONG_SEQUENCE] = apply_only_long_window, + [LONG_START_SEQUENCE] = apply_long_start_window, + [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window, + [LONG_STOP_SEQUENCE] = apply_long_stop_window +}; - if (avctx->channels > 5) - av_log(avctx, AV_LOG_ERROR, "This encoder does not yet enforce the restrictions on LFEs. " - "The output will most likely be an illegal bitstream.\n"); +static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce, + float *audio) +{ + int i; + float *output = sce->ret; - return 0; -} + apply_window[sce->ics.window_sequence[0]](&s->dsp, &s->fdsp, sce, audio); -static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s, - SingleChannelElement *sce, short *audio, int channel) -{ - int i, j, k; - const float * lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024; - const float * swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; - const float * pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; - - if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) { - memcpy(s->output, sce->saved, sizeof(float)*1024); - if (sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE) { - memset(s->output, 0, sizeof(s->output[0]) * 448); - for (i = 448; i < 576; i++) - s->output[i] = sce->saved[i] * pwindow[i - 448]; - for (i = 576; i < 704; i++) - s->output[i] = sce->saved[i]; - } - if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) { - for (i = 0, j = channel; i < 1024; i++, j += avctx->channels) { - s->output[i+1024] = audio[j] * lwindow[1024 - i - 1]; - sce->saved[i] = audio[j] * lwindow[i]; - } - } else { - for (i = 0, j = channel; i < 448; i++, j += avctx->channels) - s->output[i+1024] = audio[j]; - for (; i < 576; i++, j += avctx->channels) - s->output[i+1024] = audio[j] * swindow[576 - i - 1]; - memset(s->output+1024+576, 0, sizeof(s->output[0]) * 448); - for (i = 0, j = channel; i < 1024; i++, j += avctx->channels) - sce->saved[i] = audio[j]; - } - ff_mdct_calc(&s->mdct1024, sce->coeffs, s->output); - } else { - for (k = 0; k < 1024; k += 128) { - for (i = 448 + k; i < 448 + k + 256; i++) - s->output[i - 448 - k] = (i < 1024) - ? sce->saved[i] - : audio[channel + (i-1024)*avctx->channels]; - s->dsp.vector_fmul (s->output, k ? swindow : pwindow, 128); - s->dsp.vector_fmul_reverse(s->output+128, s->output+128, swindow, 128); - ff_mdct_calc(&s->mdct128, sce->coeffs + k, s->output); - } - for (i = 0, j = channel; i < 1024; i++, j += avctx->channels) - sce->saved[i] = audio[j]; - } + if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) + s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output); + else + for (i = 0; i < 1024; i += 128) + s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + i, output + i*2); + memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024); } /** @@ -300,10 +302,10 @@ static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe) /** * Produce integer coefficients from scalefactors provided by the model. */ -static void adjust_frame_information(AACEncContext *apc, ChannelElement *cpe, int chans) +static void adjust_frame_information(ChannelElement *cpe, int chans) { int i, w, w2, g, ch; - int start, sum, maxsfb, cmaxsfb; + int start, maxsfb, cmaxsfb; for (ch = 0; ch < chans; ch++) { IndividualChannelStream *ics = &cpe->ch[ch].ics; @@ -312,9 +314,8 @@ static void adjust_frame_information(AACEncContext *apc, ChannelElement *cpe, in cpe->ch[ch].pulse.num_pulse = 0; for (w = 0; w < ics->num_windows*16; w += 16) { for (g = 0; g < ics->num_swb; g++) { - sum = 0; //apply M/S - if (!ch && cpe->ms_mask[w + g]) { + if (cpe->common_window && !ch && cpe->ms_mask[w + g]) { for (i = 0; i < ics->swb_sizes[g]; i++) { cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) / 2.0; cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].coeffs[start+i]; @@ -356,7 +357,7 @@ static void adjust_frame_information(AACEncContext *apc, ChannelElement *cpe, in if (msc == 0 || ics0->max_sfb == 0) cpe->ms_mode = 0; else - cpe->ms_mode = msc < ics0->max_sfb ? 1 : 2; + cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2; } } @@ -459,8 +460,7 @@ static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s, /** * Write some auxiliary information about the created AAC file. */ -static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s, - const char *name) +static void put_bitstream_info(AACEncContext *s, const char *name) { int i, namelen, padbits; @@ -468,95 +468,138 @@ static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s, put_bits(&s->pb, 3, TYPE_FIL); put_bits(&s->pb, 4, FFMIN(namelen, 15)); if (namelen >= 15) - put_bits(&s->pb, 8, namelen - 16); + put_bits(&s->pb, 8, namelen - 14); put_bits(&s->pb, 4, 0); //extension type - filler - padbits = 8 - (put_bits_count(&s->pb) & 7); - align_put_bits(&s->pb); + padbits = -put_bits_count(&s->pb) & 7; + avpriv_align_put_bits(&s->pb); for (i = 0; i < namelen - 2; i++) put_bits(&s->pb, 8, name[i]); put_bits(&s->pb, 12 - padbits, 0); } -static int aac_encode_frame(AVCodecContext *avctx, - uint8_t *frame, int buf_size, void *data) +/* + * Copy input samples. + * Channels are reordered from Libav's default order to AAC order. + */ +static void copy_input_samples(AACEncContext *s, const AVFrame *frame) +{ + int ch; + int end = 2048 + (frame ? frame->nb_samples : 0); + const uint8_t *channel_map = aac_chan_maps[s->channels - 1]; + + /* copy and remap input samples */ + for (ch = 0; ch < s->channels; ch++) { + /* copy last 1024 samples of previous frame to the start of the current frame */ + memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0])); + + /* copy new samples and zero any remaining samples */ + if (frame) { + memcpy(&s->planar_samples[ch][2048], + frame->extended_data[channel_map[ch]], + frame->nb_samples * sizeof(s->planar_samples[0][0])); + } + memset(&s->planar_samples[ch][end], 0, + (3072 - end) * sizeof(s->planar_samples[0][0])); + } +} + +static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, + const AVFrame *frame, int *got_packet_ptr) { AACEncContext *s = avctx->priv_data; - int16_t *samples = s->samples, *samples2, *la; + float **samples = s->planar_samples, *samples2, *la, *overlap; ChannelElement *cpe; - int i, j, chans, tag, start_ch; - const uint8_t *chan_map = aac_chan_configs[avctx->channels-1]; + int i, ch, w, g, chans, tag, start_ch, ret; int chan_el_counter[4]; - FFPsyWindowInfo windows[avctx->channels]; + FFPsyWindowInfo windows[AAC_MAX_CHANNELS]; - if (s->last_frame) + if (s->last_frame == 2) return 0; - if (data) { - if (!s->psypp) { - memcpy(s->samples + 1024 * avctx->channels, data, - 1024 * avctx->channels * sizeof(s->samples[0])); - } else { - start_ch = 0; - samples2 = s->samples + 1024 * avctx->channels; - for (i = 0; i < chan_map[0]; i++) { - tag = chan_map[i+1]; - chans = tag == TYPE_CPE ? 2 : 1; - ff_psy_preprocess(s->psypp, (uint16_t*)data + start_ch, - samples2 + start_ch, start_ch, chans); - start_ch += chans; - } - } + + /* add current frame to queue */ + if (frame) { + if ((ret = ff_af_queue_add(&s->afq, frame) < 0)) + return ret; } - if (!avctx->frame_number) { - memcpy(s->samples, s->samples + 1024 * avctx->channels, - 1024 * avctx->channels * sizeof(s->samples[0])); + + copy_input_samples(s, frame); + if (s->psypp) + ff_psy_preprocess(s->psypp, s->planar_samples, s->channels); + + if (!avctx->frame_number) return 0; - } start_ch = 0; - for (i = 0; i < chan_map[0]; i++) { + for (i = 0; i < s->chan_map[0]; i++) { FFPsyWindowInfo* wi = windows + start_ch; - tag = chan_map[i+1]; + tag = s->chan_map[i+1]; chans = tag == TYPE_CPE ? 2 : 1; cpe = &s->cpe[i]; - samples2 = samples + start_ch; - la = samples2 + 1024 * avctx->channels + start_ch; - if (!data) - la = NULL; - for (j = 0; j < chans; j++) { - IndividualChannelStream *ics = &cpe->ch[j].ics; - int k; - wi[j] = ff_psy_suggest_window(&s->psy, samples2, la, start_ch + j, ics->window_sequence[0]); + for (ch = 0; ch < chans; ch++) { + IndividualChannelStream *ics = &cpe->ch[ch].ics; + int cur_channel = start_ch + ch; + overlap = &samples[cur_channel][0]; + samples2 = overlap + 1024; + la = samples2 + (448+64); + if (!frame) + la = NULL; + if (tag == TYPE_LFE) { + wi[ch].window_type[0] = ONLY_LONG_SEQUENCE; + wi[ch].window_shape = 0; + wi[ch].num_windows = 1; + wi[ch].grouping[0] = 1; + + /* Only the lowest 12 coefficients are used in a LFE channel. + * The expression below results in only the bottom 8 coefficients + * being used for 11.025kHz to 16kHz sample rates. + */ + ics->num_swb = s->samplerate_index >= 8 ? 1 : 3; + } else { + wi[ch] = s->psy.model->window(&s->psy, samples2, la, cur_channel, + ics->window_sequence[0]); + } ics->window_sequence[1] = ics->window_sequence[0]; - ics->window_sequence[0] = wi[j].window_type[0]; + ics->window_sequence[0] = wi[ch].window_type[0]; ics->use_kb_window[1] = ics->use_kb_window[0]; - ics->use_kb_window[0] = wi[j].window_shape; - ics->num_windows = wi[j].num_windows; + ics->use_kb_window[0] = wi[ch].window_shape; + ics->num_windows = wi[ch].num_windows; ics->swb_sizes = s->psy.bands [ics->num_windows == 8]; - ics->num_swb = s->psy.num_bands[ics->num_windows == 8]; - for (k = 0; k < ics->num_windows; k++) - ics->group_len[k] = wi[j].grouping[k]; + ics->num_swb = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8]; + for (w = 0; w < ics->num_windows; w++) + ics->group_len[w] = wi[ch].grouping[w]; - s->cur_channel = start_ch + j; - apply_window_and_mdct(avctx, s, &cpe->ch[j], samples2, j); + apply_window_and_mdct(s, &cpe->ch[ch], overlap); } start_ch += chans; } + if ((ret = ff_alloc_packet(avpkt, 768 * s->channels))) { + av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n"); + return ret; + } + do { int frame_bits; - init_put_bits(&s->pb, frame, buf_size*8); + + init_put_bits(&s->pb, avpkt->data, avpkt->size); + if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT)) - put_bitstream_info(avctx, s, LIBAVCODEC_IDENT); + put_bitstream_info(s, LIBAVCODEC_IDENT); start_ch = 0; memset(chan_el_counter, 0, sizeof(chan_el_counter)); - for (i = 0; i < chan_map[0]; i++) { + for (i = 0; i < s->chan_map[0]; i++) { FFPsyWindowInfo* wi = windows + start_ch; - tag = chan_map[i+1]; + const float *coeffs[2]; + tag = s->chan_map[i+1]; chans = tag == TYPE_CPE ? 2 : 1; cpe = &s->cpe[i]; - for (j = 0; j < chans; j++) { - s->cur_channel = start_ch + j; - ff_psy_set_band_info(&s->psy, s->cur_channel, cpe->ch[j].coeffs, &wi[j]); - s->coder->search_for_quantizers(avctx, s, &cpe->ch[j], s->lambda); + put_bits(&s->pb, 3, tag); + put_bits(&s->pb, 4, chan_el_counter[tag]++); + for (ch = 0; ch < chans; ch++) + coeffs[ch] = cpe->ch[ch].coeffs; + s->psy.model->analyze(&s->psy, start_ch, coeffs, wi); + for (ch = 0; ch < chans; ch++) { + s->cur_channel = start_ch * 2 + ch; + s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda); } cpe->common_window = 0; if (chans > 1 @@ -564,19 +607,25 @@ static int aac_encode_frame(AVCodecContext *avctx, && wi[0].window_shape == wi[1].window_shape) { cpe->common_window = 1; - for (j = 0; j < wi[0].num_windows; j++) { - if (wi[0].grouping[j] != wi[1].grouping[j]) { + for (w = 0; w < wi[0].num_windows; w++) { + if (wi[0].grouping[w] != wi[1].grouping[w]) { cpe->common_window = 0; break; } } } - s->cur_channel = start_ch; - if (cpe->common_window && s->coder->search_for_ms) - s->coder->search_for_ms(s, cpe, s->lambda); - adjust_frame_information(s, cpe, chans); - put_bits(&s->pb, 3, tag); - put_bits(&s->pb, 4, chan_el_counter[tag]++); + s->cur_channel = start_ch * 2; + if (s->options.stereo_mode && cpe->common_window) { + if (s->options.stereo_mode > 0) { + IndividualChannelStream *ics = &cpe->ch[0].ics; + for (w = 0; w < ics->num_windows; w += ics->group_len[w]) + for (g = 0; g < ics->num_swb; g++) + cpe->ms_mask[w*16+g] = 1; + } else if (s->coder->search_for_ms) { + s->coder->search_for_ms(s, cpe, s->lambda); + } + } + adjust_frame_information(cpe, chans); if (chans == 2) { put_bits(&s->pb, 1, cpe->common_window); if (cpe->common_window) { @@ -584,16 +633,18 @@ static int aac_encode_frame(AVCodecContext *avctx, encode_ms_info(&s->pb, cpe); } } - for (j = 0; j < chans; j++) { - s->cur_channel = start_ch + j; - encode_individual_channel(avctx, s, &cpe->ch[j], cpe->common_window); + for (ch = 0; ch < chans; ch++) { + s->cur_channel = start_ch + ch; + encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window); } start_ch += chans; } frame_bits = put_bits_count(&s->pb); - if (frame_bits <= 6144 * avctx->channels - 3) + if (frame_bits <= 6144 * s->channels - 3) { + s->psy.bitres.bits = frame_bits / s->channels; break; + } s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits; @@ -610,11 +661,15 @@ static int aac_encode_frame(AVCodecContext *avctx, s->lambda = FFMIN(s->lambda, 65536.f); } - if (!data) - s->last_frame = 1; - memcpy(s->samples, s->samples + 1024 * avctx->channels, - 1024 * avctx->channels * sizeof(s->samples[0])); - return put_bits_count(&s->pb)>>3; + if (!frame) + s->last_frame++; + + ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts, + &avpkt->duration); + + avpkt->size = put_bits_count(&s->pb) >> 3; + *got_packet_ptr = 1; + return 0; } static av_cold int aac_encode_end(AVCodecContext *avctx) @@ -624,21 +679,151 @@ static av_cold int aac_encode_end(AVCodecContext *avctx) ff_mdct_end(&s->mdct1024); ff_mdct_end(&s->mdct128); ff_psy_end(&s->psy); - ff_psy_preprocess_end(s->psypp); - av_freep(&s->samples); + if (s->psypp) + ff_psy_preprocess_end(s->psypp); + av_freep(&s->buffer.samples); av_freep(&s->cpe); + ff_af_queue_close(&s->afq); +#if FF_API_OLD_ENCODE_AUDIO + av_freep(&avctx->coded_frame); +#endif + return 0; +} + +static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s) +{ + int ret = 0; + + ff_dsputil_init(&s->dsp, avctx); + avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT); + + // window init + ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024); + ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128); + ff_init_ff_sine_windows(10); + ff_init_ff_sine_windows(7); + + if (ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0)) + return ret; + if (ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0)) + return ret; + + return 0; +} + +static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s) +{ + int ch; + FF_ALLOCZ_OR_GOTO(avctx, s->buffer.samples, 3 * 1024 * s->channels * sizeof(s->buffer.samples[0]), alloc_fail); + FF_ALLOCZ_OR_GOTO(avctx, s->cpe, sizeof(ChannelElement) * s->chan_map[0], alloc_fail); + FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail); + + for(ch = 0; ch < s->channels; ch++) + s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch; + +#if FF_API_OLD_ENCODE_AUDIO + if (!(avctx->coded_frame = avcodec_alloc_frame())) + goto alloc_fail; +#endif + + return 0; +alloc_fail: + return AVERROR(ENOMEM); +} + +static av_cold int aac_encode_init(AVCodecContext *avctx) +{ + AACEncContext *s = avctx->priv_data; + int i, ret = 0; + const uint8_t *sizes[2]; + uint8_t grouping[AAC_MAX_CHANNELS]; + int lengths[2]; + + avctx->frame_size = 1024; + + for (i = 0; i < 16; i++) + if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i]) + break; + + s->channels = avctx->channels; + + ERROR_IF(i == 16, + "Unsupported sample rate %d\n", avctx->sample_rate); + ERROR_IF(s->channels > AAC_MAX_CHANNELS, + "Unsupported number of channels: %d\n", s->channels); + ERROR_IF(avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW, + "Unsupported profile %d\n", avctx->profile); + ERROR_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels, + "Too many bits per frame requested\n"); + + s->samplerate_index = i; + + s->chan_map = aac_chan_configs[s->channels-1]; + + if (ret = dsp_init(avctx, s)) + goto fail; + + if (ret = alloc_buffers(avctx, s)) + goto fail; + + avctx->extradata_size = 5; + put_audio_specific_config(avctx); + + sizes[0] = swb_size_1024[i]; + sizes[1] = swb_size_128[i]; + lengths[0] = ff_aac_num_swb_1024[i]; + lengths[1] = ff_aac_num_swb_128[i]; + for (i = 0; i < s->chan_map[0]; i++) + grouping[i] = s->chan_map[i + 1] == TYPE_CPE; + if (ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping)) + goto fail; + s->psypp = ff_psy_preprocess_init(avctx); + s->coder = &ff_aac_coders[2]; + + s->lambda = avctx->global_quality ? avctx->global_quality : 120; + + ff_aac_tableinit(); + + for (i = 0; i < 428; i++) + ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i])); + + avctx->delay = 1024; + ff_af_queue_init(avctx, &s->afq); + return 0; +fail: + aac_encode_end(avctx); + return ret; } -AVCodec aac_encoder = { - "aac", - AVMEDIA_TYPE_AUDIO, - CODEC_ID_AAC, - sizeof(AACEncContext), - aac_encode_init, - aac_encode_frame, - aac_encode_end, - .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL, - .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, - .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"), +#define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM +static const AVOption aacenc_options[] = { + {"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.i64 = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"}, + {"auto", "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.i64 = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"}, + {"ms_off", "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"}, + {"ms_force", "Force Mid/Side for the whole frame if possible", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"}, + {NULL} +}; + +static const AVClass aacenc_class = { + "AAC encoder", + av_default_item_name, + aacenc_options, + LIBAVUTIL_VERSION_INT, +}; + +AVCodec ff_aac_encoder = { + .name = "aac", + .type = AVMEDIA_TYPE_AUDIO, + .id = AV_CODEC_ID_AAC, + .priv_data_size = sizeof(AACEncContext), + .init = aac_encode_init, + .encode2 = aac_encode_frame, + .close = aac_encode_end, + .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | + CODEC_CAP_EXPERIMENTAL, + .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP, + AV_SAMPLE_FMT_NONE }, + .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"), + .priv_class = &aacenc_class, };