#include <lame/lame.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"
#include "avcodec.h"
+#include "audio_frame_queue.h"
#include "internal.h"
#include "mpegaudio.h"
#include "mpegaudiodecheader.h"
typedef struct LAMEContext {
AVClass *class;
+ AVCodecContext *avctx;
lame_global_flags *gfp;
- uint8_t buffer[BUFFER_SIZE];
+ uint8_t *buffer;
int buffer_index;
+ int buffer_size;
int reservoir;
+ 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) {
+ int new_size = s->buffer_index + 2 * BUFFER_SIZE, err;
+
+ av_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size,
+ new_size);
+ if ((err = av_reallocp(&s->buffer, new_size)) < 0) {
+ s->buffer_size = s->buffer_index = 0;
+ return err;
+ }
+ s->buffer_size = new_size;
+ }
+ return 0;
+}
+
static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
{
LAMEContext *s = avctx->priv_data;
- av_freep(&avctx->coded_frame);
+ av_freep(&s->samples_flt[0]);
+ av_freep(&s->samples_flt[1]);
+ av_freep(&s->buffer);
+
+ ff_af_queue_close(&s->afq);
lame_close(s->gfp);
return 0;
LAMEContext *s = avctx->priv_data;
int ret;
+ s->avctx = avctx;
+
/* initialize LAME and get defaults */
if ((s->gfp = lame_init()) == NULL)
return AVERROR(ENOMEM);
- /* channels */
- if (avctx->channels > 2) {
- ret = AVERROR(EINVAL);
- goto error;
- }
lame_set_num_channels(s->gfp, avctx->channels);
lame_set_mode(s->gfp, avctx->channels > 1 ? JOINT_STEREO : MONO);
goto error;
}
+ /* get encoder delay */
+ avctx->delay = lame_get_encoder_delay(s->gfp) + 528 + 1;
+ ff_af_queue_init(avctx, &s->afq);
+
avctx->frame_size = lame_get_framesize(s->gfp);
- avctx->coded_frame = avcodec_alloc_frame();
- if (!avctx->coded_frame) {
- ret = AVERROR(ENOMEM);
- goto error;
+
+ /* allocate float sample buffers */
+ if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
+ int ch;
+ for (ch = 0; ch < avctx->channels; 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;
}
-static int mp3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
- int buf_size, void *data)
+#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 mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+ const AVFrame *frame, int *got_packet_ptr)
{
LAMEContext *s = avctx->priv_data;
MPADecodeHeader hdr;
- int len;
+ int len, ret, ch;
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);
+ if (frame) {
+ switch (avctx->sample_fmt) {
+ case AV_SAMPLE_FMT_S16P:
+ ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
+ break;
+ case AV_SAMPLE_FMT_S32P:
+ ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
+ break;
+ 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)
+ return ret;
+ }
/* Move 1 frame from the LAME buffer to the output packet, if available.
We have to parse the first frame header in the output buffer to
av_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
s->buffer_index);
if (len <= s->buffer_index) {
- memcpy(frame, s->buffer, len);
+ if ((ret = ff_alloc_packet(avpkt, len))) {
+ av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
+ return ret;
+ }
+ memcpy(avpkt->data, s->buffer, len);
s->buffer_index -= len;
memmove(s->buffer, s->buffer + len, s->buffer_index);
- return len;
- } else
- return 0;
+
+ /* Get the next frame pts/duration */
+ ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
+ &avpkt->duration);
+
+ avpkt->size = len;
+ *got_packet_ptr = 1;
+ }
+ return 0;
}
#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 },
};
AVCodec ff_libmp3lame_encoder = {
.name = "libmp3lame",
+ .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
.type = AVMEDIA_TYPE_AUDIO,
- .id = CODEC_ID_MP3,
+ .id = AV_CODEC_ID_MP3,
.priv_data_size = sizeof(LAMEContext),
.init = mp3lame_encode_init,
- .encode = mp3lame_encode_frame,
+ .encode2 = mp3lame_encode_frame,
.close = mp3lame_encode_close,
- .capabilities = CODEC_CAP_DELAY,
- .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
+ .capabilities = CODEC_CAP_DELAY | CODEC_CAP_SMALL_LAST_FRAME,
+ .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,
- .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
+ .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
+ AV_CH_LAYOUT_STEREO,
+ 0 },
.priv_class = &libmp3lame_class,
.defaults = libmp3lame_defaults,
};