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author | Justin Ruggles <justin.ruggles@gmail.com> | 2012-08-25 13:44:30 -0400 |
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committer | Justin Ruggles <justin.ruggles@gmail.com> | 2012-10-06 13:23:13 -0400 |
commit | 31b2262dca9cc77709d20c45610ec8030e7f9257 (patch) | |
tree | 4a4d18370b4755a04e2f6d0d2ca5c4e49c2146cf /libavcodec/wmaenc.c | |
parent | b1540fc8843dfe35b228c733d7aeca2e83ef4739 (diff) | |
download | ffmpeg-31b2262dca9cc77709d20c45610ec8030e7f9257.tar.gz |
wmaenc: use float planar sample format
Diffstat (limited to 'libavcodec/wmaenc.c')
-rw-r--r-- | libavcodec/wmaenc.c | 32 |
1 files changed, 16 insertions, 16 deletions
diff --git a/libavcodec/wmaenc.c b/libavcodec/wmaenc.c index b439548829..8abb55bbd4 100644 --- a/libavcodec/wmaenc.c +++ b/libavcodec/wmaenc.c @@ -97,23 +97,24 @@ static int encode_init(AVCodecContext * avctx){ } -static void apply_window_and_mdct(AVCodecContext * avctx, const signed short * audio, int len) { +static void apply_window_and_mdct(AVCodecContext * avctx, const AVFrame *frame) +{ WMACodecContext *s = avctx->priv_data; + float **audio = (float **)frame->extended_data; + int len = frame->nb_samples; int window_index= s->frame_len_bits - s->block_len_bits; FFTContext *mdct = &s->mdct_ctx[window_index]; - int i, j, channel; + int ch; const float * win = s->windows[window_index]; int window_len = 1 << s->block_len_bits; - float n = window_len/2; - - for (channel = 0; channel < avctx->channels; channel++) { - memcpy(s->output, s->frame_out[channel], sizeof(float)*window_len); - j = channel; - for (i = 0; i < len; i++, j += avctx->channels){ - s->output[i+window_len] = audio[j] / n * win[window_len - i - 1]; - s->frame_out[channel][i] = audio[j] / n * win[i]; - } - mdct->mdct_calc(mdct, s->coefs[channel], s->output); + float n = 2.0 * 32768.0 / window_len; + + for (ch = 0; ch < avctx->channels; ch++) { + memcpy(s->output, s->frame_out[ch], window_len * sizeof(*s->output)); + s->dsp.vector_fmul_scalar(s->frame_out[ch], audio[ch], n, len); + s->dsp.vector_fmul_reverse(&s->output[window_len], s->frame_out[ch], win, len); + s->fdsp.vector_fmul(s->frame_out[ch], s->frame_out[ch], win, len); + mdct->mdct_calc(mdct, s->coefs[ch], s->output); } } @@ -349,13 +350,12 @@ static int encode_superframe(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr) { WMACodecContext *s = avctx->priv_data; - const int16_t *samples = (const int16_t *)frame->data[0]; int i, total_gain, ret; s->block_len_bits= s->frame_len_bits; //required by non variable block len s->block_len = 1 << s->block_len_bits; - apply_window_and_mdct(avctx, samples, frame->nb_samples); + apply_window_and_mdct(avctx, frame); if (s->ms_stereo) { float a, b; @@ -426,7 +426,7 @@ AVCodec ff_wmav1_encoder = { .init = encode_init, .encode2 = encode_superframe, .close = ff_wma_end, - .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, + .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE }, .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 1"), }; @@ -439,7 +439,7 @@ AVCodec ff_wmav2_encoder = { .init = encode_init, .encode2 = encode_superframe, .close = ff_wma_end, - .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, + .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE }, .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 2"), }; |