diff options
| author | Tim Redfern <tim@eclectronics.org> | 2013-12-29 12:19:38 +0000 |
|---|---|---|
| committer | Tim Redfern <tim@eclectronics.org> | 2013-12-29 12:19:38 +0000 |
| commit | f7813a5324be39d13ab536c245d15dfc602a7849 (patch) | |
| tree | fad99148b88823d34a5df2f0a25881a002eb291b /ffmpeg/libavcodec/aacenc.c | |
| parent | b7a5a477b8ff4d4e3028b9dfb9a9df0a41463f92 (diff) | |
basic type mechanism working
Diffstat (limited to 'ffmpeg/libavcodec/aacenc.c')
| -rw-r--r-- | ffmpeg/libavcodec/aacenc.c | 12 |
1 files changed, 8 insertions, 4 deletions
diff --git a/ffmpeg/libavcodec/aacenc.c b/ffmpeg/libavcodec/aacenc.c index 80dd3d8..5596b4b 100644 --- a/ffmpeg/libavcodec/aacenc.c +++ b/ffmpeg/libavcodec/aacenc.c @@ -593,7 +593,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, 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->cur_channel = start_ch + ch; s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda); } cpe->common_window = 0; @@ -609,7 +609,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, } } } - s->cur_channel = start_ch * 2; + s->cur_channel = start_ch; if (s->options.stereo_mode && cpe->common_window) { if (s->options.stereo_mode > 0) { IndividualChannelStream *ics = &cpe->ch[0].ics; @@ -791,7 +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 = 2}, 0, AAC_CODER_NB-1, AACENC_FLAGS}, + {"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} }; @@ -811,6 +815,7 @@ static const int mpeg4audio_sample_rates[16] = { AVCodec ff_aac_encoder = { .name = "aac", + .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"), .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_AAC, .priv_data_size = sizeof(AACEncContext), @@ -822,6 +827,5 @@ AVCodec ff_aac_encoder = { 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, }; |