* Interface to libmp3lame for mp3 encoding
* Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
- * @file libavcodec/libmp3lame.c
+ * @file
* Interface to libmp3lame for mp3 encoding.
*/
+#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 <lame/lame.h>
+#include "mpegaudiodecheader.h"
-#define BUFFER_SIZE (7200 + 2*MPA_FRAME_SIZE + MPA_FRAME_SIZE/4)
-typedef struct Mp3AudioContext {
+#define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4)
+
+typedef struct LAMEContext {
+ AVClass *class;
+ AVCodecContext *avctx;
lame_global_flags *gfp;
- int stereo;
- uint8_t buffer[BUFFER_SIZE];
+ uint8_t *buffer;
int buffer_index;
-} Mp3AudioContext;
+ int buffer_size;
+ int reservoir;
+ int joint_stereo;
+ int abr;
+ float *samples_flt[2];
+ AudioFrameQueue afq;
+ AVFloatDSPContext fdsp;
+} LAMEContext;
-static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
-{
- Mp3AudioContext *s = avctx->priv_data;
- if (avctx->channels > 2)
- return -1;
-
- s->stereo = avctx->channels > 1 ? 1 : 0;
-
- if ((s->gfp = lame_init()) == NULL)
- goto err;
- lame_set_in_samplerate(s->gfp, avctx->sample_rate);
- lame_set_out_samplerate(s->gfp, avctx->sample_rate);
- lame_set_num_channels(s->gfp, avctx->channels);
- if(avctx->compression_level == FF_COMPRESSION_DEFAULT) {
- lame_set_quality(s->gfp, 5);
- } else {
- lame_set_quality(s->gfp, avctx->compression_level);
- }
- /* lame 3.91 doesn't work in mono */
- lame_set_mode(s->gfp, JOINT_STEREO);
- lame_set_brate(s->gfp, avctx->bit_rate/1000);
- if(avctx->flags & CODEC_FLAG_QSCALE) {
- lame_set_brate(s->gfp, 0);
- lame_set_VBR(s->gfp, vbr_default);
- lame_set_VBR_q(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
+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;
+
+ ff_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;
}
- lame_set_bWriteVbrTag(s->gfp,0);
- lame_set_disable_reservoir(s->gfp, avctx->flags2 & CODEC_FLAG2_BIT_RESERVOIR ? 0 : 1);
- if (lame_init_params(s->gfp) < 0)
- goto err_close;
+ return 0;
+}
- avctx->frame_size = lame_get_framesize(s->gfp);
+static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
+{
+ LAMEContext *s = avctx->priv_data;
- avctx->coded_frame= avcodec_alloc_frame();
- avctx->coded_frame->key_frame= 1;
+ av_freep(&s->samples_flt[0]);
+ av_freep(&s->samples_flt[1]);
+ av_freep(&s->buffer);
- return 0;
+ ff_af_queue_close(&s->afq);
-err_close:
lame_close(s->gfp);
-err:
- return -1;
+ return 0;
}
-static const int sSampleRates[3] = {
- 44100, 48000, 32000
-};
+static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
+{
+ LAMEContext *s = avctx->priv_data;
+ int ret;
-static const int sBitRates[2][3][15] = {
- { { 0, 32, 64, 96,128,160,192,224,256,288,320,352,384,416,448},
- { 0, 32, 48, 56, 64, 80, 96,112,128,160,192,224,256,320,384},
- { 0, 32, 40, 48, 56, 64, 80, 96,112,128,160,192,224,256,320}
- },
- { { 0, 32, 48, 56, 64, 80, 96,112,128,144,160,176,192,224,256},
- { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160},
- { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160}
- },
-};
+ s->avctx = avctx;
-static const int sSamplesPerFrame[2][3] =
-{
- { 384, 1152, 1152 },
- { 384, 1152, 576 }
-};
+ /* initialize LAME and get defaults */
+ if (!(s->gfp = lame_init()))
+ return AVERROR(ENOMEM);
-static const int sBitsPerSlot[3] = {
- 32,
- 8,
- 8
-};
+ lame_set_num_channels(s->gfp, avctx->channels);
+ lame_set_mode(s->gfp, avctx->channels > 1 ? s->joint_stereo ? JOINT_STEREO : STEREO : MONO);
-static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
-{
- uint32_t header = AV_RB32(data);
- int layerID = 3 - ((header >> 17) & 0x03);
- int bitRateID = ((header >> 12) & 0x0f);
- int sampleRateID = ((header >> 10) & 0x03);
- int bitsPerSlot = sBitsPerSlot[layerID];
- int isPadded = ((header >> 9) & 0x01);
- static int const mode_tab[4]= {2,3,1,0};
- int mode= mode_tab[(header >> 19) & 0x03];
- int mpeg_id= mode>0;
- int temp0, temp1, bitRate;
-
- if ( (( header >> 21 ) & 0x7ff) != 0x7ff || mode == 3 || layerID==3 || sampleRateID==3) {
- return -1;
+ /* sample rate */
+ lame_set_in_samplerate (s->gfp, avctx->sample_rate);
+ lame_set_out_samplerate(s->gfp, avctx->sample_rate);
+
+ /* algorithmic quality */
+ if (avctx->compression_level == FF_COMPRESSION_DEFAULT)
+ lame_set_quality(s->gfp, 5);
+ else
+ lame_set_quality(s->gfp, avctx->compression_level);
+
+ /* rate control */
+ if (avctx->flags & AV_CODEC_FLAG_QSCALE) { // VBR
+ lame_set_VBR(s->gfp, vbr_default);
+ lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
+ } else {
+ if (avctx->bit_rate) {
+ if (s->abr) { // ABR
+ lame_set_VBR(s->gfp, vbr_abr);
+ lame_set_VBR_mean_bitrate_kbps(s->gfp, avctx->bit_rate / 1000);
+ } else // CBR
+ lame_set_brate(s->gfp, avctx->bit_rate / 1000);
+ }
+ }
+
+ /* do not get a Xing VBR header frame from LAME */
+ lame_set_bWriteVbrTag(s->gfp,0);
+
+ /* bit reservoir usage */
+ lame_set_disable_reservoir(s->gfp, !s->reservoir);
+
+ /* set specified parameters */
+ if (lame_init_params(s->gfp) < 0) {
+ ret = -1;
+ goto error;
}
- if(!samplesPerFrame) samplesPerFrame= &temp0;
- if(!sampleRate ) sampleRate = &temp1;
+ /* get encoder delay */
+ avctx->initial_padding = lame_get_encoder_delay(s->gfp) + 528 + 1;
+ ff_af_queue_init(avctx, &s->afq);
+
+ avctx->frame_size = lame_get_framesize(s->gfp);
+
+ /* 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;
+ }
+ }
+ }
-// *isMono = ((header >> 6) & 0x03) == 0x03;
+ ret = realloc_buffer(s);
+ if (ret < 0)
+ goto error;
- *sampleRate = sSampleRates[sampleRateID]>>mode;
- bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
- *samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID];
-//av_log(NULL, AV_LOG_DEBUG, "sr:%d br:%d spf:%d l:%d m:%d\n", *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
+ avpriv_float_dsp_init(&s->fdsp, avctx->flags & AV_CODEC_FLAG_BITEXACT);
- return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
+ 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)
{
- Mp3AudioContext *s = avctx->priv_data;
- int len;
+ LAMEContext *s = avctx->priv_data;
+ MPADecodeHeader hdr;
+ int len, ret, ch;
int lame_result;
-
- /* lame 3.91 dies on '1-channel interleaved' data */
-
- if(data){
- if (s->stereo) {
- 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
- );
+ uint32_t h;
+
+ 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
- );
+ } else {
+ lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
+ s->buffer_size - s->buffer_index);
}
-
- if(lame_result < 0){
- if(lame_result==-1) {
- /* output buffer too small */
- 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);
+ 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, 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;
+ }
- if(s->buffer_index<4)
- return 0;
+ /* add current frame to the queue */
+ if (frame) {
+ if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
+ return ret;
+ }
- len= mp3len(s->buffer, NULL, NULL);
-//av_log(avctx, AV_LOG_DEBUG, "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);
+ /* 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. */
+ if (s->buffer_index < 4)
+ return 0;
+ h = AV_RB32(s->buffer);
+
+ ret = avpriv_mpegaudio_decode_header(&hdr, h);
+ if (ret < 0) {
+ av_log(avctx, AV_LOG_ERROR, "Invalid mp3 header at start of buffer\n");
+ return AVERROR_BUG;
+ } else if (ret) {
+ av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
+ return -1;
+ }
+ len = hdr.frame_size;
+ ff_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
+ s->buffer_index);
+ if (len <= s->buffer_index) {
+ 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);
- memmove(s->buffer, s->buffer+len, s->buffer_index);
- //FIXME fix the audio codec API, so we do not need the memcpy()
-/*for(i=0; i<len; i++){
- av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
-}*/
- 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;
}
-static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
-{
- Mp3AudioContext *s = avctx->priv_data;
+#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, { .i64 = 1 }, 0, 1, AE },
+ { "joint_stereo", "Use joint stereo.", OFFSET(joint_stereo), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE },
+ { "abr", "Use ABR", OFFSET(abr), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AE },
+ { NULL },
+};
- av_freep(&avctx->coded_frame);
+static const AVClass libmp3lame_class = {
+ .class_name = "libmp3lame encoder",
+ .item_name = av_default_item_name,
+ .option = options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
- lame_close(s->gfp);
- return 0;
-}
+static const AVCodecDefault libmp3lame_defaults[] = {
+ { "b", "0" },
+ { NULL },
+};
+static const int libmp3lame_sample_rates[] = {
+ 44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
+};
-AVCodec libmp3lame_encoder = {
- "libmp3lame",
- CODEC_TYPE_AUDIO,
- CODEC_ID_MP3,
- sizeof(Mp3AudioContext),
- MP3lame_encode_init,
- MP3lame_encode_frame,
- MP3lame_encode_close,
- .capabilities= CODEC_CAP_DELAY,
- .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
- .long_name= NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
+AVCodec ff_libmp3lame_encoder = {
+ .name = "libmp3lame",
+ .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_MP3,
+ .priv_data_size = sizeof(LAMEContext),
+ .init = mp3lame_encode_init,
+ .encode2 = mp3lame_encode_frame,
+ .close = mp3lame_encode_close,
+ .capabilities = AV_CODEC_CAP_DELAY | AV_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,
+ .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
+ AV_CH_LAYOUT_STEREO,
+ 0 },
+ .priv_class = &libmp3lame_class,
+ .defaults = libmp3lame_defaults,
};
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