#include <lame/lame.h>
-#include "libavutil/audioconvert.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/common.h"
+#include "libavutil/float_dsp.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/log.h"
#include "libavutil/opt.h"
AVClass *class;
AVCodecContext *avctx;
lame_global_flags *gfp;
- uint8_t buffer[BUFFER_SIZE];
+ uint8_t *buffer;
int buffer_index;
+ int buffer_size;
int reservoir;
- void *planar_samples[2];
+ float *samples_flt[2];
AudioFrameQueue afq;
+ AVFloatDSPContext fdsp;
} LAMEContext;
+static int realloc_buffer(LAMEContext *s)
+{
+ if (!s->buffer || s->buffer_size - s->buffer_index < BUFFER_SIZE) {
+ uint8_t *tmp;
+ int new_size = s->buffer_index + 2 * BUFFER_SIZE;
+
+ av_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size,
+ new_size);
+ tmp = av_realloc(s->buffer, new_size);
+ if (!tmp) {
+ av_freep(&s->buffer);
+ s->buffer_size = s->buffer_index = 0;
+ return AVERROR(ENOMEM);
+ }
+ s->buffer = tmp;
+ s->buffer_size = new_size;
+ }
+ return 0;
+}
+
static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
{
LAMEContext *s = avctx->priv_data;
-#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]);
+ av_freep(&s->buffer);
ff_af_queue_close(&s->afq);
avctx->frame_size = lame_get_framesize(s->gfp);
-#if FF_API_OLD_ENCODE_AUDIO
- avctx->coded_frame = avcodec_alloc_frame();
- if (!avctx->coded_frame) {
- ret = AVERROR(ENOMEM);
- goto error;
- }
-#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;
}
}
}
+ ret = realloc_buffer(s);
+ if (ret < 0)
+ goto error;
+
+ avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
+
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, \
+ s->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->fdsp.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;
}
} else {
lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
- BUFFER_SIZE - s->buffer_index);
+ s->buffer_size - s->buffer_index);
}
if (lame_result < 0) {
if (lame_result == -1) {
av_log(avctx, AV_LOG_ERROR,
"lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
- s->buffer_index, BUFFER_SIZE - s->buffer_index);
+ s->buffer_index, s->buffer_size - s->buffer_index);
}
return -1;
}
s->buffer_index += lame_result;
+ ret = realloc_buffer(s);
+ if (ret < 0) {
+ av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n");
+ return ret;
+ }
/* add current frame to the queue */
if (frame) {
- if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
+ if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
return ret;
}
#define OFFSET(x) offsetof(LAMEContext, x)
#define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
static const AVOption options[] = {
- { "reservoir", "Use bit reservoir.", OFFSET(reservoir), AV_OPT_TYPE_INT, { 1 }, 0, 1, AE },
+ { "reservoir", "Use bit reservoir.", OFFSET(reservoir), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE },
{ NULL },
};
.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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