diff options
Diffstat (limited to 'libavcodec/aacenc.c')
| -rw-r--r-- | libavcodec/aacenc.c | 57 |
1 files changed, 38 insertions, 19 deletions
diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c index a28e1a1290..aa6a56a128 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 Libav. + * This file is part of FFmpeg. * - * Libav is free software; you can redistribute it and/or + * 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. * - * Libav is distributed in the hope that it will be useful, + * 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 Libav; if not, write to the Free Software + * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -145,7 +145,7 @@ static const uint8_t aac_chan_configs[6][5] = { }; /** - * Table to remap channels from Libav's default order to AAC order. + * Table to remap channels from libavcodec's default order to AAC order. */ static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = { { 0 }, @@ -252,7 +252,7 @@ static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce, int i; float *output = sce->ret_buf; - apply_window[sce->ics.window_sequence[0]](&s->fdsp, sce, audio); + apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, audio); if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output); @@ -384,8 +384,7 @@ static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s, for (i = 0; i < sce->ics.max_sfb; i++) { if (!sce->zeroes[w*16 + i]) { diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO; - if (diff < 0 || diff > 120) - av_log(avctx, AV_LOG_ERROR, "Scalefactor difference is too big to be coded\n"); + av_assert0(diff >= 0 && diff <= 120); off = sce->sf_idx[w*16 + i]; put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]); } @@ -478,7 +477,7 @@ static void put_bitstream_info(AACEncContext *s, const char *name) /* * Copy input samples. - * Channels are reordered from Libav's default order to AAC order. + * Channels are reordered from libavcodec's default order to AAC order. */ static void copy_input_samples(AACEncContext *s, const AVFrame *frame) { @@ -568,14 +567,15 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, ics->group_len[w] = wi[ch].grouping[w]; apply_window_and_mdct(s, &cpe->ch[ch], overlap); + if (isnan(cpe->ch->coeffs[0])) { + av_log(avctx, AV_LOG_ERROR, "Input contains NaN\n"); + return AVERROR(EINVAL); + } } start_ch += chans; } - if ((ret = ff_alloc_packet(avpkt, 768 * s->channels))) { - av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n"); + if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels)) < 0) return ret; - } - do { int frame_bits; @@ -682,6 +682,7 @@ static av_cold int aac_encode_end(AVCodecContext *avctx) ff_psy_preprocess_end(s->psypp); av_freep(&s->buffer.samples); av_freep(&s->cpe); + av_freep(&s->fdsp); ff_af_queue_close(&s->afq); return 0; } @@ -690,7 +691,9 @@ static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s) { int ret = 0; - avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT); + s->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT); + if (!s->fdsp) + return AVERROR(ENOMEM); // window init ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024); @@ -709,8 +712,8 @@ static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s) 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_ARRAY_OR_GOTO(avctx, s->buffer.samples, s->channels, 3 * 1024 * sizeof(s->buffer.samples[0]), alloc_fail); + FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->cpe, s->chan_map[0], sizeof(ChannelElement), alloc_fail); FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail); for(ch = 0; ch < s->channels; ch++) @@ -737,7 +740,7 @@ static av_cold int aac_encode_init(AVCodecContext *avctx) s->channels = avctx->channels; - ERROR_IF(i == 16, + ERROR_IF(i >= 12, "Unsupported sample rate %d\n", avctx->sample_rate); ERROR_IF(s->channels > AAC_MAX_CHANNELS, "Unsupported number of channels: %d\n", s->channels); @@ -769,9 +772,12 @@ static av_cold int aac_encode_init(AVCodecContext *avctx) s->chan_map[0], grouping)) < 0) goto fail; s->psypp = ff_psy_preprocess_init(avctx); - s->coder = &ff_aac_coders[2]; + s->coder = &ff_aac_coders[s->options.aac_coder]; - s->lambda = avctx->global_quality ? avctx->global_quality : 120; + if (HAVE_MIPSDSPR1) + ff_aac_coder_init_mips(s); + + s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120; ff_aac_tableinit(); @@ -793,6 +799,11 @@ static const AVOption aacenc_options[] = { {"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"}, + {"aac_coder", "", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "aac_coder"}, + {"faac", "FAAC-inspired method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAAC}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"}, + {"anmr", "ANMR method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"}, + {"twoloop", "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"}, + {"fast", "Constant quantizer", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"}, {NULL} }; @@ -803,6 +814,13 @@ static const AVClass aacenc_class = { LIBAVUTIL_VERSION_INT, }; +/* duplicated from avpriv_mpeg4audio_sample_rates to avoid shared build + * failures */ +static const int mpeg4audio_sample_rates[16] = { + 96000, 88200, 64000, 48000, 44100, 32000, + 24000, 22050, 16000, 12000, 11025, 8000, 7350 +}; + AVCodec ff_aac_encoder = { .name = "aac", .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"), @@ -812,6 +830,7 @@ AVCodec ff_aac_encoder = { .init = aac_encode_init, .encode2 = aac_encode_frame, .close = aac_encode_end, + .supported_samplerates = mpeg4audio_sample_rates, .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL, .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP, |