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author | Justin Ruggles <justin.ruggles@gmail.com> | 2012-08-21 09:21:56 -0400 |
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committer | Justin Ruggles <justin.ruggles@gmail.com> | 2012-08-22 16:41:41 -0400 |
commit | f24cc1b2f1b5d9a1fd91340cc9f7b97dda9dc3b1 (patch) | |
tree | cac8032865688d3ae07f8a4f5d065deef650e8ae /libavcodec | |
parent | 358078d9bb89d6266e274720eba8582ec7b1c6b7 (diff) | |
download | ffmpeg-streaming-f24cc1b2f1b5d9a1fd91340cc9f7b97dda9dc3b1.zip ffmpeg-streaming-f24cc1b2f1b5d9a1fd91340cc9f7b97dda9dc3b1.tar.gz |
alacenc: use s16p sample format as input
Diffstat (limited to 'libavcodec')
-rw-r--r-- | libavcodec/alacenc.c | 25 |
1 files changed, 12 insertions, 13 deletions
diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c index d337e40..6b5c4f0 100644 --- a/libavcodec/alacenc.c +++ b/libavcodec/alacenc.c @@ -78,17 +78,15 @@ typedef struct AlacEncodeContext { } AlacEncodeContext; -static void init_sample_buffers(AlacEncodeContext *s, - const int16_t *input_samples) +static void init_sample_buffers(AlacEncodeContext *s, int16_t **input_samples) { int ch, i; for (ch = 0; ch < s->avctx->channels; ch++) { - const int16_t *sptr = input_samples + ch; - for (i = 0; i < s->frame_size; i++) { - s->sample_buf[ch][i] = *sptr; - sptr += s->avctx->channels; - } + int32_t *bptr = s->sample_buf[ch]; + const int16_t *sptr = input_samples[ch]; + for (i = 0; i < s->frame_size; i++) + bptr[i] = sptr[i]; } } @@ -347,8 +345,7 @@ static void alac_entropy_coder(AlacEncodeContext *s) } } -static int write_frame(AlacEncodeContext *s, AVPacket *avpkt, - const int16_t *samples) +static int write_frame(AlacEncodeContext *s, AVPacket *avpkt, int16_t **samples) { int i, j; int prediction_type = 0; @@ -358,8 +355,10 @@ static int write_frame(AlacEncodeContext *s, AVPacket *avpkt, if (s->verbatim) { write_frame_header(s); - for (i = 0; i < s->frame_size * s->avctx->channels; i++) - put_sbits(pb, 16, *samples++); + /* samples are channel-interleaved in verbatim mode */ + for (i = 0; i < s->frame_size; i++) + for (j = 0; j < s->avctx->channels; j++) + put_sbits(pb, 16, samples[j][i]); } else { init_sample_buffers(s, samples); write_frame_header(s); @@ -537,7 +536,7 @@ static int alac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, { AlacEncodeContext *s = avctx->priv_data; int out_bytes, max_frame_size, ret; - const int16_t *samples = (const int16_t *)frame->data[0]; + int16_t **samples = (int16_t **)frame->extended_data; s->frame_size = frame->nb_samples; @@ -577,7 +576,7 @@ AVCodec ff_alac_encoder = { .encode2 = alac_encode_frame, .close = alac_encode_close, .capabilities = CODEC_CAP_SMALL_LAST_FRAME, - .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, + .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_NONE }, .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"), }; |