1/*
2 * Interface to libmp3lame for mp3 encoding
3 * Copyright (c) 2002 Lennert Buytenhek <buytenh@gnu.org>
4 *
5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22/**
23 * @file
24 * Interface to libmp3lame for mp3 encoding.
25 */
26
27#include <lame/lame.h>
28
29#include "libavutil/channel_layout.h"
30#include "libavutil/common.h"
31#include "libavutil/float_dsp.h"
32#include "libavutil/intreadwrite.h"
33#include "libavutil/log.h"
34#include "libavutil/opt.h"
35#include "avcodec.h"
36#include "audio_frame_queue.h"
37#include "codec_internal.h"
38#include "encode.h"
39#include "mpegaudio.h"
40#include "mpegaudiodecheader.h"
41
42#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.
43
44typedef struct LAMEContext {
45    AVClass *class;
46    AVCodecContext *avctx;
47    lame_global_flags *gfp;
48    uint8_t *buffer;
49    int buffer_index;
50    int buffer_size;
51    int reservoir;
52    int joint_stereo;
53    int abr;
54    int delay_sent;
55    float *samples_flt[2];
56    AudioFrameQueue afq;
57    AVFloatDSPContext *fdsp;
58} LAMEContext;
59
60
61static int realloc_buffer(LAMEContext *s)
62{
63    if (!s->buffer || s->buffer_size - s->buffer_index < BUFFER_SIZE) {
64        int new_size = s->buffer_index + 2 * BUFFER_SIZE, err;
65
66        ff_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size,
67                new_size);
68        if ((err = av_reallocp(&s->buffer, new_size)) < 0) {
69            s->buffer_size = s->buffer_index = 0;
70            return err;
71        }
72        s->buffer_size = new_size;
73    }
74    return 0;
75}
76
77static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
78{
79    LAMEContext *s = avctx->priv_data;
80
81    av_freep(&s->samples_flt[0]);
82    av_freep(&s->samples_flt[1]);
83    av_freep(&s->buffer);
84    av_freep(&s->fdsp);
85
86    ff_af_queue_close(&s->afq);
87
88    lame_close(s->gfp);
89    return 0;
90}
91
92static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
93{
94    LAMEContext *s = avctx->priv_data;
95    int ret;
96
97    s->avctx = avctx;
98
99    /* initialize LAME and get defaults */
100    if (!(s->gfp = lame_init()))
101        return AVERROR(ENOMEM);
102
103
104    lame_set_num_channels(s->gfp, avctx->ch_layout.nb_channels);
105    lame_set_mode(s->gfp, avctx->ch_layout.nb_channels > 1 ?
106                          s->joint_stereo ? JOINT_STEREO : STEREO : MONO);
107
108    /* sample rate */
109    lame_set_in_samplerate (s->gfp, avctx->sample_rate);
110    lame_set_out_samplerate(s->gfp, avctx->sample_rate);
111
112    /* algorithmic quality */
113    if (avctx->compression_level != FF_COMPRESSION_DEFAULT)
114        lame_set_quality(s->gfp, avctx->compression_level);
115
116    /* rate control */
117    if (avctx->flags & AV_CODEC_FLAG_QSCALE) { // VBR
118        lame_set_VBR(s->gfp, vbr_default);
119        lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
120    } else {
121        if (avctx->bit_rate) {
122            if (s->abr) {                   // ABR
123                lame_set_VBR(s->gfp, vbr_abr);
124                lame_set_VBR_mean_bitrate_kbps(s->gfp, avctx->bit_rate / 1000);
125            } else                          // CBR
126                lame_set_brate(s->gfp, avctx->bit_rate / 1000);
127        }
128    }
129
130    /* lowpass cutoff frequency */
131    if (avctx->cutoff)
132        lame_set_lowpassfreq(s->gfp, avctx->cutoff);
133
134    /* do not get a Xing VBR header frame from LAME */
135    lame_set_bWriteVbrTag(s->gfp,0);
136
137    /* bit reservoir usage */
138    lame_set_disable_reservoir(s->gfp, !s->reservoir);
139
140    /* set specified parameters */
141    if (lame_init_params(s->gfp) < 0) {
142        ret = AVERROR_EXTERNAL;
143        goto error;
144    }
145
146    /* get encoder delay */
147    avctx->initial_padding = lame_get_encoder_delay(s->gfp) + 528 + 1;
148    ff_af_queue_init(avctx, &s->afq);
149
150    avctx->frame_size  = lame_get_framesize(s->gfp);
151
152    /* allocate float sample buffers */
153    if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) {
154        int ch;
155        for (ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
156            s->samples_flt[ch] = av_malloc_array(avctx->frame_size,
157                                           sizeof(*s->samples_flt[ch]));
158            if (!s->samples_flt[ch]) {
159                ret = AVERROR(ENOMEM);
160                goto error;
161            }
162        }
163    }
164
165    ret = realloc_buffer(s);
166    if (ret < 0)
167        goto error;
168
169    s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
170    if (!s->fdsp) {
171        ret = AVERROR(ENOMEM);
172        goto error;
173    }
174
175
176    return 0;
177error:
178    mp3lame_encode_close(avctx);
179    return ret;
180}
181
182#define ENCODE_BUFFER(func, buf_type, buf_name) do {                        \
183    lame_result = func(s->gfp,                                              \
184                       (const buf_type *)buf_name[0],                       \
185                       (const buf_type *)buf_name[1], frame->nb_samples,    \
186                       s->buffer + s->buffer_index,                         \
187                       s->buffer_size - s->buffer_index);                   \
188} while (0)
189
190static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
191                                const AVFrame *frame, int *got_packet_ptr)
192{
193    LAMEContext *s = avctx->priv_data;
194    MPADecodeHeader hdr;
195    int len, ret, ch, discard_padding;
196    int lame_result;
197    uint32_t h;
198
199    if (frame) {
200        switch (avctx->sample_fmt) {
201        case AV_SAMPLE_FMT_S16P:
202            ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data);
203            break;
204        case AV_SAMPLE_FMT_S32P:
205            ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data);
206            break;
207        case AV_SAMPLE_FMT_FLTP:
208            if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) {
209                av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n");
210                return AVERROR(EINVAL);
211            }
212            for (ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
213                s->fdsp->vector_fmul_scalar(s->samples_flt[ch],
214                                           (const float *)frame->data[ch],
215                                           32768.0f,
216                                           FFALIGN(frame->nb_samples, 8));
217            }
218            ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt);
219            break;
220        default:
221            return AVERROR_BUG;
222        }
223    } else if (!s->afq.frame_alloc) {
224        lame_result = 0;
225    } else {
226        lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
227                                        s->buffer_size - s->buffer_index);
228    }
229    if (lame_result < 0) {
230        if (lame_result == -1) {
231            av_log(avctx, AV_LOG_ERROR,
232                   "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
233                   s->buffer_index, s->buffer_size - s->buffer_index);
234        }
235        return AVERROR(ENOMEM);
236    }
237    s->buffer_index += lame_result;
238    ret = realloc_buffer(s);
239    if (ret < 0) {
240        av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n");
241        return ret;
242    }
243
244    /* add current frame to the queue */
245    if (frame) {
246        if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
247            return ret;
248    }
249
250    /* Move 1 frame from the LAME buffer to the output packet, if available.
251       We have to parse the first frame header in the output buffer to
252       determine the frame size. */
253    if (s->buffer_index < 4)
254        return 0;
255    h = AV_RB32(s->buffer);
256
257    ret = avpriv_mpegaudio_decode_header(&hdr, h);
258    if (ret < 0) {
259        av_log(avctx, AV_LOG_ERROR, "Invalid mp3 header at start of buffer\n");
260        return AVERROR_BUG;
261    } else if (ret) {
262        av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
263        return AVERROR_INVALIDDATA;
264    }
265    len = hdr.frame_size;
266    ff_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
267            s->buffer_index);
268    if (len <= s->buffer_index) {
269        if ((ret = ff_get_encode_buffer(avctx, avpkt, len, 0)) < 0)
270            return ret;
271        memcpy(avpkt->data, s->buffer, len);
272        s->buffer_index -= len;
273        memmove(s->buffer, s->buffer + len, s->buffer_index);
274
275        /* Get the next frame pts/duration */
276        ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
277                           &avpkt->duration);
278
279        discard_padding = avctx->frame_size - avpkt->duration;
280        // Check if subtraction resulted in an overflow
281        if ((discard_padding < avctx->frame_size) != (avpkt->duration > 0)) {
282            av_log(avctx, AV_LOG_ERROR, "discard padding overflow\n");
283            av_packet_unref(avpkt);
284            return AVERROR(EINVAL);
285        }
286        if ((!s->delay_sent && avctx->initial_padding > 0) || discard_padding > 0) {
287            uint8_t* side_data = av_packet_new_side_data(avpkt,
288                                                         AV_PKT_DATA_SKIP_SAMPLES,
289                                                         10);
290            if(!side_data) {
291                av_packet_unref(avpkt);
292                return AVERROR(ENOMEM);
293            }
294            if (!s->delay_sent) {
295                AV_WL32(side_data, avctx->initial_padding);
296                s->delay_sent = 1;
297            }
298            AV_WL32(side_data + 4, discard_padding);
299        }
300
301        *got_packet_ptr = 1;
302    }
303    return 0;
304}
305
306#define OFFSET(x) offsetof(LAMEContext, x)
307#define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
308static const AVOption options[] = {
309    { "reservoir",    "use bit reservoir", OFFSET(reservoir),    AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE },
310    { "joint_stereo", "use joint stereo",  OFFSET(joint_stereo), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AE },
311    { "abr",          "use ABR",           OFFSET(abr),          AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, AE },
312    { NULL },
313};
314
315static const AVClass libmp3lame_class = {
316    .class_name = "libmp3lame encoder",
317    .item_name  = av_default_item_name,
318    .option     = options,
319    .version    = LIBAVUTIL_VERSION_INT,
320};
321
322static const FFCodecDefault libmp3lame_defaults[] = {
323    { "b",          "0" },
324    { NULL },
325};
326
327static const int libmp3lame_sample_rates[] = {
328    44100, 48000,  32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
329};
330
331const FFCodec ff_libmp3lame_encoder = {
332    .p.name                = "libmp3lame",
333    .p.long_name           = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
334    .p.type                = AVMEDIA_TYPE_AUDIO,
335    .p.id                  = AV_CODEC_ID_MP3,
336    .p.capabilities        = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY |
337                             AV_CODEC_CAP_SMALL_LAST_FRAME,
338    .priv_data_size        = sizeof(LAMEContext),
339    .init                  = mp3lame_encode_init,
340    FF_CODEC_ENCODE_CB(mp3lame_encode_frame),
341    .close                 = mp3lame_encode_close,
342    .p.sample_fmts         = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P,
343                                                             AV_SAMPLE_FMT_FLTP,
344                                                             AV_SAMPLE_FMT_S16P,
345                                                             AV_SAMPLE_FMT_NONE },
346    .p.supported_samplerates = libmp3lame_sample_rates,
347#if FF_API_OLD_CHANNEL_LAYOUT
348    .p.channel_layouts     = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
349                                                  AV_CH_LAYOUT_STEREO,
350                                                  0 },
351#endif
352    .p.ch_layouts          = (const AVChannelLayout[]) { AV_CHANNEL_LAYOUT_MONO,
353                                                         AV_CHANNEL_LAYOUT_STEREO,
354                                                         { 0 },
355    },
356    .p.priv_class          = &libmp3lame_class,
357    .defaults              = libmp3lame_defaults,
358    .p.wrapper_name        = "libmp3lame",
359};
360