#include <lame/lame.h>
+#include "libavutil/audioconvert.h"
+#include "libavutil/common.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];
int buffer_index;
int reservoir;
+ void *planar_samples[2];
+ AudioFrameQueue afq;
} LAMEContext;
{
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]);
+
+ 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);
+
+#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) {
+ 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:
return ret;
}
-static int mp3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
- int buf_size, void *data)
+#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; \
+ } \
+ } \
+} 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;
+ int len, ret;
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_S16:
+ lame_result = encode_frame_int16(s, frame->data[0], frame->nb_samples);
+ break;
+ case AV_SAMPLE_FMT_S32:
+ lame_result = encode_frame_int32(s, frame->data[0], frame->nb_samples);
+ break;
+ case AV_SAMPLE_FMT_FLT:
+ lame_result = encode_frame_float(s, frame->data[0], frame->nb_samples);
+ break;
+ default:
+ return AVERROR_BUG;
}
} else {
lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
}
s->buffer_index += lame_result;
+ /* 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
determine the frame size. */
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",
.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_S32,
+ AV_SAMPLE_FMT_FLT,
+ AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.supported_samplerates = libmp3lame_sample_rates,
+ .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
+ AV_CH_LAYOUT_STEREO,
+ 0 },
.long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
.priv_class = &libmp3lame_class,
.defaults = libmp3lame_defaults,
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