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-rw-r--r--libavcodec/aacenc.c40
1 files changed, 26 insertions, 14 deletions
diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c
index 55aa2f1a2f..499aefb9fe 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 },
@@ -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)
{
@@ -571,11 +570,8 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
}
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;
@@ -768,9 +764,12 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
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->coder = &ff_aac_coders[s->options.aac_coder];
+
+ if (HAVE_MIPSDSPR1)
+ ff_aac_coder_init_mips(s);
- s->lambda = avctx->global_quality ? avctx->global_quality : 120;
+ s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120;
ff_aac_tableinit();
@@ -792,6 +791,11 @@ 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}
};
@@ -802,6 +806,13 @@ 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)"),
@@ -811,6 +822,7 @@ 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,