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Code Issues Releases Activity

libmp3lame: support float and s32 sample formats

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This commit is contained in:
Justin Ruggles 2012-02-17 01:50:57 -05:00
parent e232225276
commit e00959a9b1
1 changed files with 84 additions and 11 deletions
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95
libavcodec/libmp3lame.c
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@ -38,10 +38,12 @@
typedef struct LAMEContext {
AVClass *class;
AVCodecContext *avctx;
lame_global_flags *gfp;
uint8_t buffer[BUFFER_SIZE];
int buffer_index;
int reservoir;
void *planar_samples[2];
} LAMEContext;
@ -50,6 +52,8 @@ static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
LAMEContext *s = avctx->priv_data;
av_freep(&avctx->coded_frame);
av_freep(&s->planar_samples[0]);
av_freep(&s->planar_samples[1]);
lame_close(s->gfp);
return 0;
@ -60,6 +64,8 @@ static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
LAMEContext *s = avctx->priv_data;
int ret;
s->avctx = avctx;
/* initialize LAME and get defaults */
if ((s->gfp = lame_init()) == NULL)
return AVERROR(ENOMEM);
@ -110,12 +116,75 @@ static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
goto error;
}
/* sample format */
if (avctx->sample_fmt == AV_SAMPLE_FMT_S32 ||
avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
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]) {
ret = AVERROR(ENOMEM);
goto error;
}
}
}
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 < s->avctx->frame_size; i++) { \
output[i] = *input * scale; \
input += s->avctx->channels; \
} \
} \
} while (0)
static int encode_frame_int16(LAMEContext *s, void *samples)
{
if (s->avctx->channels > 1) {
return lame_encode_buffer_interleaved(s->gfp, samples,
s->avctx->frame_size,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index);
} else {
return lame_encode_buffer(s->gfp, samples, NULL, s->avctx->frame_size,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index);
}
}
static int encode_frame_int32(LAMEContext *s, void *samples)
{
DEINTERLEAVE(int32_t, 1);
return lame_encode_buffer_int(s->gfp,
s->planar_samples[0], s->planar_samples[1],
s->avctx->frame_size,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index);
}
static int encode_frame_float(LAMEContext *s, void *samples)
{
DEINTERLEAVE(float, 32768.0f);
return lame_encode_buffer_float(s->gfp,
s->planar_samples[0], s->planar_samples[1],
s->avctx->frame_size,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index);
}
static int mp3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
int buf_size, void *data)
{
@ -125,16 +194,18 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
int lame_result;
if (data) {
if (avctx->channels > 1) {
lame_result = lame_encode_buffer_interleaved(s->gfp, data,
avctx->frame_size,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index);
} else {
lame_result = lame_encode_buffer(s->gfp, data, data,
avctx->frame_size, s->buffer +
s->buffer_index, BUFFER_SIZE -
s->buffer_index);
switch (avctx->sample_fmt) {
case AV_SAMPLE_FMT_S16:
lame_result = encode_frame_int16(s, data);
break;
case AV_SAMPLE_FMT_S32:
lame_result = encode_frame_int32(s, data);
break;
case AV_SAMPLE_FMT_FLT:
lame_result = encode_frame_float(s, data);
break;
default:
return AVERROR_BUG;
}
} else {
lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
@ -203,7 +274,9 @@ AVCodec ff_libmp3lame_encoder = {
.encode = mp3lame_encode_frame,
.close = mp3lame_encode_close,
.capabilities = CODEC_CAP_DELAY,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.supported_samplerates = libmp3lame_sample_rates,
.long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),