FFmpeg/libavcodec/libmp3lame.c
wm4 b945fed629 avcodec: add metadata to identify wrappers and hardware decoders
Explicitly identify decoder/encoder wrappers with a common name. This
saves API users from guessing by the name suffix. For example, they
don't have to guess that "h264_qsv" is the h264 QSV implementation, and
instead they can just check the AVCodec .codec and .wrapper_name fields.

Explicitly mark AVCodec entries that are hardware decoders or most
likely hardware decoders with new AV_CODEC_CAPs. The purpose is allowing
API users listing hardware decoders in a more generic way. The proposed
AVCodecHWConfig does not provide this information fully, because it's
concerned with decoder configuration, not information about the fact
whether the hardware is used or not.

AV_CODEC_CAP_HYBRID exists specifically for QSV, which can have software
implementations in case the hardware is not capable.

Based on a patch by Philip Langdale <philipl@overt.org>.

Merges Libav commit 47687a2f8a.
2017-12-14 19:37:56 +01:00

354 lines
12 KiB
C

/*
* Interface to libmp3lame for mp3 encoding
* Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
*
* This file is part of FFmpeg.
*
* 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.
*
* 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @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 "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.
typedef struct LAMEContext {
AVClass *class;
AVCodecContext *avctx;
lame_global_flags *gfp;
uint8_t *buffer;
int buffer_index;
int buffer_size;
int reservoir;
int joint_stereo;
int abr;
int delay_sent;
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;
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;
}
return 0;
}
static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
{
LAMEContext *s = avctx->priv_data;
av_freep(&s->samples_flt[0]);
av_freep(&s->samples_flt[1]);
av_freep(&s->buffer);
av_freep(&s->fdsp);
ff_af_queue_close(&s->afq);
lame_close(s->gfp);
return 0;
}
static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
{
LAMEContext *s = avctx->priv_data;
int ret;
s->avctx = avctx;
/* initialize LAME and get defaults */
if (!(s->gfp = lame_init()))
return AVERROR(ENOMEM);
lame_set_num_channels(s->gfp, avctx->channels);
lame_set_mode(s->gfp, avctx->channels > 1 ? s->joint_stereo ? JOINT_STEREO : 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, 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);
}
}
/* lowpass cutoff frequency */
if (avctx->cutoff)
lame_set_lowpassfreq(s->gfp, avctx->cutoff);
/* 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;
}
/* 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_array(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;
s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
if (!s->fdsp) {
ret = AVERROR(ENOMEM);
goto error;
}
return 0;
error:
mp3lame_encode_close(avctx);
return ret;
}
#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, ret, ch, discard_padding;
int lame_result;
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 if (!s->afq.frame_alloc) {
lame_result = 0;
} 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) {
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;
}
/* 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;
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_packet2(avctx, avpkt, len, 0)) < 0)
return ret;
memcpy(avpkt->data, s->buffer, len);
s->buffer_index -= len;
memmove(s->buffer, s->buffer + len, s->buffer_index);
/* Get the next frame pts/duration */
ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
&avpkt->duration);
discard_padding = avctx->frame_size - avpkt->duration;
// Check if subtraction resulted in an overflow
if ((discard_padding < avctx->frame_size) != (avpkt->duration > 0)) {
av_log(avctx, AV_LOG_ERROR, "discard padding overflow\n");
av_packet_unref(avpkt);
av_free(avpkt);
return AVERROR(EINVAL);
}
if ((!s->delay_sent && avctx->initial_padding > 0) || discard_padding > 0) {
uint8_t* side_data = av_packet_new_side_data(avpkt,
AV_PKT_DATA_SKIP_SAMPLES,
10);
if(!side_data) {
av_packet_unref(avpkt);
av_free(avpkt);
return AVERROR(ENOMEM);
}
if (!s->delay_sent) {
AV_WL32(side_data, avctx->initial_padding);
s->delay_sent = 1;
}
AV_WL32(side_data + 4, discard_padding);
}
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_BOOL, { .i64 = 1 }, 0, 1, AE },
{ "joint_stereo", "use joint stereo", OFFSET(joint_stereo), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE },
{ "abr", "use ABR", OFFSET(abr), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, AE },
{ NULL },
};
static const AVClass libmp3lame_class = {
.class_name = "libmp3lame encoder",
.item_name = av_default_item_name,
.option = options,
.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",
.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,
.wrapper_name = "libmp3lame",
};