Commit 473b297f authored by Justin Ruggles's avatar Justin Ruggles
Browse files

libmp3lame: use planar sample formats

parent 233783e2
......@@ -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 +128,63 @@ 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(func, buf_type, buf_name) do { \
lame_result = func(s->gfp, \
(const buf_type *)buf_name[0], \
(const buf_type *)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(lame_encode_buffer, 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(lame_encode_buffer_int, 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(lame_encode_buffer_float, float, s->samples_flt);
break;
default:
return AVERROR_BUG;
......@@ -300,9 +272,9 @@ 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,
......
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