---
libavcodec/libmp3lame.c | 105 +++++++++++++++++-----------------------------
1 files changed, 39 insertions(+), 66 deletions(-)
diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c
index d754f46..b09b61c 100644
--- a/libavcodec/libmp3lame.c
+++ b/libavcodec/libmp3lame.c
@@ -33,6 +33,7 @@
#include "libavutil/opt.h"
#include "avcodec.h"
#include "audio_frame_queue.h"
+#include "dsputil.h"
#include "internal.h"
#include "mpegaudio.h"
#include "mpegaudiodecheader.h"
@@ -46,8 +47,9 @@ typedef struct LAMEContext {
uint8_t buffer[BUFFER_SIZE];
int buffer_index;
int reservoir;
- void *planar_samples[2];
+ float *samples_flt[2];
AudioFrameQueue afq;
+ DSPContext dsp;
} LAMEContext;
@@ -58,8 +60,8 @@ static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
- av_freep(&s->planar_samples[0]);
- av_freep(&s->planar_samples[1]);
+ av_freep(&s->samples_flt[0]);
+ av_freep(&s->samples_flt[1]);
ff_af_queue_close(&s->afq);
@@ -126,93 +128,64 @@ static av_cold int mp3lame_encode_init(AVCodecContext
*avctx)
}
#endif
- /* sample format */
- if (avctx->sample_fmt == AV_SAMPLE_FMT_S32 ||
- avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
+ /* allocate float sample buffers */
+ if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
int ch;
for (ch = 0; ch < avctx->channels; ch++) {
- s->planar_samples[ch] = av_malloc(avctx->frame_size *
-
av_get_bytes_per_sample(avctx->sample_fmt));
- if (!s->planar_samples[ch]) {
+ s->samples_flt[ch] = av_malloc(avctx->frame_size *
+ sizeof(*s->samples_flt[ch]));
+ if (!s->samples_flt[ch]) {
ret = AVERROR(ENOMEM);
goto error;
}
}
}
+ ff_dsputil_init(&s->dsp, avctx);
+
return 0;
error:
mp3lame_encode_close(avctx);
return ret;
}
-#define DEINTERLEAVE(type, scale) do { \
- int ch, i; \
- for (ch = 0; ch < s->avctx->channels; ch++) { \
- const type *input = samples; \
- type *output = s->planar_samples[ch]; \
- input += ch; \
- for (i = 0; i < nb_samples; i++) { \
- output[i] = *input * scale; \
- input += s->avctx->channels; \
- } \
- } \
+#define ENCODE_BUFFER(suffix, buf_name) do { \
+ lame_result = lame_encode_ ## suffix(s->gfp, \
+ buf_name[0], \
+ buf_name[1], \
+ frame->nb_samples, \
+ s->buffer + s->buffer_index, \
+ BUFFER_SIZE - s->buffer_index); \
} while (0)
-static int encode_frame_int16(LAMEContext *s, void *samples, int nb_samples)
-{
- if (s->avctx->channels > 1) {
- return lame_encode_buffer_interleaved(s->gfp, samples,
- nb_samples,
- s->buffer + s->buffer_index,
- BUFFER_SIZE - s->buffer_index);
- } else {
- return lame_encode_buffer(s->gfp, samples, NULL, nb_samples,
- s->buffer + s->buffer_index,
- BUFFER_SIZE - s->buffer_index);
- }
-}
-
-static int encode_frame_int32(LAMEContext *s, void *samples, int nb_samples)
-{
- DEINTERLEAVE(int32_t, 1);
-
- return lame_encode_buffer_int(s->gfp,
- s->planar_samples[0], s->planar_samples[1],
- nb_samples,
- s->buffer + s->buffer_index,
- BUFFER_SIZE - s->buffer_index);
-}
-
-static int encode_frame_float(LAMEContext *s, void *samples, int nb_samples)
-{
- DEINTERLEAVE(float, 32768.0f);
-
- return lame_encode_buffer_float(s->gfp,
- s->planar_samples[0], s->planar_samples[1],
- nb_samples,
- s->buffer + s->buffer_index,
- BUFFER_SIZE - s->buffer_index);
-}
-
static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
LAMEContext *s = avctx->priv_data;
MPADecodeHeader hdr;
- int len, ret;
+ int len, ret, ch;
int lame_result;
if (frame) {
switch (avctx->sample_fmt) {
- case AV_SAMPLE_FMT_S16:
- lame_result = encode_frame_int16(s, frame->data[0],
frame->nb_samples);
+ case AV_SAMPLE_FMT_S16P:
+ ENCODE_BUFFER(buffer, (const int16_t *)frame->data);
break;
- case AV_SAMPLE_FMT_S32:
- lame_result = encode_frame_int32(s, frame->data[0],
frame->nb_samples);
+ case AV_SAMPLE_FMT_S32P:
+ ENCODE_BUFFER(buffer_int, (const int32_t *)frame->data);
break;
- case AV_SAMPLE_FMT_FLT:
- lame_result = encode_frame_float(s, frame->data[0],
frame->nb_samples);
+ case AV_SAMPLE_FMT_FLTP:
+ if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
+ av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane
padding\n");
+ return AVERROR(EINVAL);
+ }
+ for (ch = 0; ch < avctx->channels; ch++) {
+ s->dsp.vector_fmul_scalar(s->samples_flt[ch],
+ (const float *)frame->data[ch],
+ 32768.0f,
+ FFALIGN(frame->nb_samples, 8));
+ }
+ ENCODE_BUFFER(buffer_float, s->samples_flt);
break;
default:
return AVERROR_BUG;
@@ -300,9 +273,9 @@ AVCodec ff_libmp3lame_encoder = {
.encode2 = mp3lame_encode_frame,
.close = mp3lame_encode_close,
.capabilities = CODEC_CAP_DELAY | CODEC_CAP_SMALL_LAST_FRAME,
- .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32,
- AV_SAMPLE_FMT_FLT,
- AV_SAMPLE_FMT_S16,
+ .sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_S32P,
+
AV_SAMPLE_FMT_FLTP,
+
AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE },
.supported_samplerates = libmp3lame_sample_rates,
.channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
--
1.7.1
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