* Interface to libmp3lame for mp3 encoding
* Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg 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.
*
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg 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 Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
* Interface to libmp3lame for mp3 encoding.
*/
+#include <lame/lame.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+1000) // FIXME: Buffer size to small? Adding 1000 to make up for it.
-#define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4)
-typedef struct Mp3AudioContext {
+typedef struct LAMEContext {
AVClass *class;
+ AVCodecContext *avctx;
lame_global_flags *gfp;
uint8_t buffer[BUFFER_SIZE];
int buffer_index;
int reservoir;
-} Mp3AudioContext;
+ void *planar_samples[2];
+ AudioFrameQueue afq;
+} LAMEContext;
-static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
+static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
{
- Mp3AudioContext *s = avctx->priv_data;
+ 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;
}
-static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
+static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
{
- Mp3AudioContext *s = avctx->priv_data;
+ LAMEContext *s = avctx->priv_data;
int ret;
- if (avctx->channels > 2)
- return AVERROR(EINVAL);
+ s->avctx = avctx;
+ /* initialize LAME and get defaults */
if ((s->gfp = lame_init()) == NULL)
return AVERROR(ENOMEM);
- lame_set_in_samplerate(s->gfp, avctx->sample_rate);
- lame_set_out_samplerate(s->gfp, avctx->sample_rate);
+
+ /* channels */
+ if (avctx->channels > 2) {
+ av_log(avctx, AV_LOG_ERROR,
+ "Invalid number of channels %d, must be <= 2\n", avctx->channels);
+ ret = AVERROR(EINVAL);
+ goto error;
+ }
lame_set_num_channels(s->gfp, avctx->channels);
- if (avctx->compression_level == FF_COMPRESSION_DEFAULT) {
+ lame_set_mode(s->gfp, avctx->channels > 1 ? JOINT_STEREO : MONO);
+
+ /* 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 {
+ else
lame_set_quality(s->gfp, avctx->compression_level);
- }
- lame_set_mode(s->gfp, avctx->channels > 1 ? JOINT_STEREO : MONO);
- lame_set_brate(s->gfp, avctx->bit_rate / 1000);
+
+ /* rate control */
if (avctx->flags & CODEC_FLAG_QSCALE) {
- lame_set_brate(s->gfp, 0);
lame_set_VBR(s->gfp, vbr_default);
lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
+ } else {
+ if (avctx->bit_rate)
+ 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;
}
- avctx->frame_size = lame_get_framesize(s->gfp);
- avctx->coded_frame = avcodec_alloc_frame();
+ /* 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:
- MP3lame_encode_close(avctx);
+ mp3lame_encode_close(avctx);
return ret;
}
-static const int sSampleRates[] = {
- 44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
-};
-
-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 }
- },
-};
-
-static const int sSamplesPerFrame[2][3] = {
- { 384, 1152, 1152 },
- { 384, 1152, 576 }
-};
-
-static const int sBitsPerSlot[3] = { 32, 8, 8 };
-
-static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
+#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)
{
- 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;
+ 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);
}
+}
- if (!samplesPerFrame)
- samplesPerFrame = &temp0;
- if (!sampleRate)
- sampleRate = &temp1;
+static int encode_frame_int32(LAMEContext *s, void *samples, int nb_samples)
+{
+ DEINTERLEAVE(int32_t, 1);
- //*isMono = ((header >> 6) & 0x03) == 0x03;
+ 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);
+}
- *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);
+static int encode_frame_float(LAMEContext *s, void *samples, int nb_samples)
+{
+ DEINTERLEAVE(float, 32768.0f);
- return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
+ 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, unsigned char *frame,
- int buf_size, void *data)
+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;
int lame_result;
- /* lame 3.91 dies on '1-channel interleaved' data */
-
- 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,
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);
}
return -1;
}
-
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. */
if (s->buffer_index < 4)
return 0;
-
- 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 (avpriv_mpegaudio_decode_header(&hdr, AV_RB32(s->buffer))) {
+ av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
+ return -1;
+ }
+ len = hdr.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_packet2(avctx, avpkt, len)))
+ return ret;
+ memcpy(avpkt->data, s->buffer, len);
s->buffer_index -= len;
-
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;
}
-#define OFFSET(x) offsetof(Mp3AudioContext, x)
+#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 },
.version = LIBAVUTIL_VERSION_INT,
};
+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 ff_libmp3lame_encoder = {
.name = "libmp3lame",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_MP3,
- .priv_data_size = sizeof(Mp3AudioContext),
- .init = MP3lame_encode_init,
- .encode = MP3lame_encode_frame,
- .close = MP3lame_encode_close,
- .capabilities = CODEC_CAP_DELAY,
- .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
+ .priv_data_size = sizeof(LAMEContext),
+ .init = mp3lame_encode_init,
+ .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,
AV_SAMPLE_FMT_NONE },
- .supported_samplerates = sSampleRates,
+ .supported_samplerates = libmp3lame_sample_rates,
.long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
.priv_class = &libmp3lame_class,
+ .defaults = libmp3lame_defaults,
};