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2ba45a60 DM |
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 "internal.h" | |
38 | #include "mpegaudio.h" | |
39 | #include "mpegaudiodecheader.h" | |
40 | ||
41 | #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. | |
42 | ||
43 | typedef struct LAMEContext { | |
44 | AVClass *class; | |
45 | AVCodecContext *avctx; | |
46 | lame_global_flags *gfp; | |
47 | uint8_t *buffer; | |
48 | int buffer_index; | |
49 | int buffer_size; | |
50 | int reservoir; | |
51 | int joint_stereo; | |
52 | int abr; | |
53 | float *samples_flt[2]; | |
54 | AudioFrameQueue afq; | |
f6fa7814 | 55 | AVFloatDSPContext *fdsp; |
2ba45a60 DM |
56 | } LAMEContext; |
57 | ||
58 | ||
59 | static int realloc_buffer(LAMEContext *s) | |
60 | { | |
61 | if (!s->buffer || s->buffer_size - s->buffer_index < BUFFER_SIZE) { | |
62 | int new_size = s->buffer_index + 2 * BUFFER_SIZE, err; | |
63 | ||
64 | av_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size, | |
65 | new_size); | |
66 | if ((err = av_reallocp(&s->buffer, new_size)) < 0) { | |
67 | s->buffer_size = s->buffer_index = 0; | |
68 | return err; | |
69 | } | |
70 | s->buffer_size = new_size; | |
71 | } | |
72 | return 0; | |
73 | } | |
74 | ||
75 | static av_cold int mp3lame_encode_close(AVCodecContext *avctx) | |
76 | { | |
77 | LAMEContext *s = avctx->priv_data; | |
78 | ||
79 | av_freep(&s->samples_flt[0]); | |
80 | av_freep(&s->samples_flt[1]); | |
81 | av_freep(&s->buffer); | |
f6fa7814 | 82 | av_freep(&s->fdsp); |
2ba45a60 DM |
83 | |
84 | ff_af_queue_close(&s->afq); | |
85 | ||
86 | lame_close(s->gfp); | |
87 | return 0; | |
88 | } | |
89 | ||
90 | static av_cold int mp3lame_encode_init(AVCodecContext *avctx) | |
91 | { | |
92 | LAMEContext *s = avctx->priv_data; | |
93 | int ret; | |
94 | ||
95 | s->avctx = avctx; | |
96 | ||
97 | /* initialize LAME and get defaults */ | |
98 | if (!(s->gfp = lame_init())) | |
99 | return AVERROR(ENOMEM); | |
100 | ||
101 | ||
102 | lame_set_num_channels(s->gfp, avctx->channels); | |
103 | lame_set_mode(s->gfp, avctx->channels > 1 ? s->joint_stereo ? JOINT_STEREO : STEREO : MONO); | |
104 | ||
105 | /* sample rate */ | |
106 | lame_set_in_samplerate (s->gfp, avctx->sample_rate); | |
107 | lame_set_out_samplerate(s->gfp, avctx->sample_rate); | |
108 | ||
109 | /* algorithmic quality */ | |
f6fa7814 | 110 | if (avctx->compression_level != FF_COMPRESSION_DEFAULT) |
2ba45a60 DM |
111 | lame_set_quality(s->gfp, avctx->compression_level); |
112 | ||
113 | /* rate control */ | |
114 | if (avctx->flags & CODEC_FLAG_QSCALE) { // VBR | |
115 | lame_set_VBR(s->gfp, vbr_default); | |
116 | lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA); | |
117 | } else { | |
118 | if (avctx->bit_rate) { | |
119 | if (s->abr) { // ABR | |
120 | lame_set_VBR(s->gfp, vbr_abr); | |
121 | lame_set_VBR_mean_bitrate_kbps(s->gfp, avctx->bit_rate / 1000); | |
122 | } else // CBR | |
123 | lame_set_brate(s->gfp, avctx->bit_rate / 1000); | |
124 | } | |
125 | } | |
126 | ||
127 | /* do not get a Xing VBR header frame from LAME */ | |
128 | lame_set_bWriteVbrTag(s->gfp,0); | |
129 | ||
130 | /* bit reservoir usage */ | |
131 | lame_set_disable_reservoir(s->gfp, !s->reservoir); | |
132 | ||
133 | /* set specified parameters */ | |
134 | if (lame_init_params(s->gfp) < 0) { | |
135 | ret = -1; | |
136 | goto error; | |
137 | } | |
138 | ||
139 | /* get encoder delay */ | |
f6fa7814 | 140 | avctx->initial_padding = lame_get_encoder_delay(s->gfp) + 528 + 1; |
2ba45a60 DM |
141 | ff_af_queue_init(avctx, &s->afq); |
142 | ||
143 | avctx->frame_size = lame_get_framesize(s->gfp); | |
144 | ||
145 | /* allocate float sample buffers */ | |
146 | if (avctx->sample_fmt == AV_SAMPLE_FMT_FLTP) { | |
147 | int ch; | |
148 | for (ch = 0; ch < avctx->channels; ch++) { | |
149 | s->samples_flt[ch] = av_malloc(avctx->frame_size * | |
150 | sizeof(*s->samples_flt[ch])); | |
151 | if (!s->samples_flt[ch]) { | |
152 | ret = AVERROR(ENOMEM); | |
153 | goto error; | |
154 | } | |
155 | } | |
156 | } | |
157 | ||
158 | ret = realloc_buffer(s); | |
159 | if (ret < 0) | |
160 | goto error; | |
161 | ||
f6fa7814 DM |
162 | s->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT); |
163 | if (!s->fdsp) { | |
164 | ret = AVERROR(ENOMEM); | |
165 | goto error; | |
166 | } | |
167 | ||
2ba45a60 DM |
168 | |
169 | return 0; | |
170 | error: | |
171 | mp3lame_encode_close(avctx); | |
172 | return ret; | |
173 | } | |
174 | ||
175 | #define ENCODE_BUFFER(func, buf_type, buf_name) do { \ | |
176 | lame_result = func(s->gfp, \ | |
177 | (const buf_type *)buf_name[0], \ | |
178 | (const buf_type *)buf_name[1], frame->nb_samples, \ | |
179 | s->buffer + s->buffer_index, \ | |
180 | s->buffer_size - s->buffer_index); \ | |
181 | } while (0) | |
182 | ||
183 | static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, | |
184 | const AVFrame *frame, int *got_packet_ptr) | |
185 | { | |
186 | LAMEContext *s = avctx->priv_data; | |
187 | MPADecodeHeader hdr; | |
188 | int len, ret, ch; | |
189 | int lame_result; | |
190 | uint32_t h; | |
191 | ||
192 | if (frame) { | |
193 | switch (avctx->sample_fmt) { | |
194 | case AV_SAMPLE_FMT_S16P: | |
195 | ENCODE_BUFFER(lame_encode_buffer, int16_t, frame->data); | |
196 | break; | |
197 | case AV_SAMPLE_FMT_S32P: | |
198 | ENCODE_BUFFER(lame_encode_buffer_int, int32_t, frame->data); | |
199 | break; | |
200 | case AV_SAMPLE_FMT_FLTP: | |
201 | if (frame->linesize[0] < 4 * FFALIGN(frame->nb_samples, 8)) { | |
202 | av_log(avctx, AV_LOG_ERROR, "inadequate AVFrame plane padding\n"); | |
203 | return AVERROR(EINVAL); | |
204 | } | |
205 | for (ch = 0; ch < avctx->channels; ch++) { | |
f6fa7814 | 206 | s->fdsp->vector_fmul_scalar(s->samples_flt[ch], |
2ba45a60 DM |
207 | (const float *)frame->data[ch], |
208 | 32768.0f, | |
209 | FFALIGN(frame->nb_samples, 8)); | |
210 | } | |
211 | ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt); | |
212 | break; | |
213 | default: | |
214 | return AVERROR_BUG; | |
215 | } | |
216 | } else if (!s->afq.frame_alloc) { | |
217 | lame_result = 0; | |
218 | } else { | |
219 | lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index, | |
220 | s->buffer_size - s->buffer_index); | |
221 | } | |
222 | if (lame_result < 0) { | |
223 | if (lame_result == -1) { | |
224 | av_log(avctx, AV_LOG_ERROR, | |
225 | "lame: output buffer too small (buffer index: %d, free bytes: %d)\n", | |
226 | s->buffer_index, s->buffer_size - s->buffer_index); | |
227 | } | |
228 | return -1; | |
229 | } | |
230 | s->buffer_index += lame_result; | |
231 | ret = realloc_buffer(s); | |
232 | if (ret < 0) { | |
233 | av_log(avctx, AV_LOG_ERROR, "error reallocating output buffer\n"); | |
234 | return ret; | |
235 | } | |
236 | ||
237 | /* add current frame to the queue */ | |
238 | if (frame) { | |
239 | if ((ret = ff_af_queue_add(&s->afq, frame)) < 0) | |
240 | return ret; | |
241 | } | |
242 | ||
243 | /* Move 1 frame from the LAME buffer to the output packet, if available. | |
244 | We have to parse the first frame header in the output buffer to | |
245 | determine the frame size. */ | |
246 | if (s->buffer_index < 4) | |
247 | return 0; | |
248 | h = AV_RB32(s->buffer); | |
249 | if (ff_mpa_check_header(h) < 0) { | |
250 | av_log(avctx, AV_LOG_ERROR, "Invalid mp3 header at start of buffer\n"); | |
251 | return AVERROR_BUG; | |
252 | } | |
253 | if (avpriv_mpegaudio_decode_header(&hdr, h)) { | |
254 | av_log(avctx, AV_LOG_ERROR, "free format output not supported\n"); | |
255 | return -1; | |
256 | } | |
257 | len = hdr.frame_size; | |
258 | av_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, | |
259 | s->buffer_index); | |
260 | if (len <= s->buffer_index) { | |
261 | if ((ret = ff_alloc_packet2(avctx, avpkt, len)) < 0) | |
262 | return ret; | |
263 | memcpy(avpkt->data, s->buffer, len); | |
264 | s->buffer_index -= len; | |
265 | memmove(s->buffer, s->buffer + len, s->buffer_index); | |
266 | ||
267 | /* Get the next frame pts/duration */ | |
268 | ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts, | |
269 | &avpkt->duration); | |
270 | ||
271 | avpkt->size = len; | |
272 | *got_packet_ptr = 1; | |
273 | } | |
274 | return 0; | |
275 | } | |
276 | ||
277 | #define OFFSET(x) offsetof(LAMEContext, x) | |
278 | #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM | |
279 | static const AVOption options[] = { | |
280 | { "reservoir", "use bit reservoir", OFFSET(reservoir), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE }, | |
281 | { "joint_stereo", "use joint stereo", OFFSET(joint_stereo), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AE }, | |
282 | { "abr", "use ABR", OFFSET(abr), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AE }, | |
283 | { NULL }, | |
284 | }; | |
285 | ||
286 | static const AVClass libmp3lame_class = { | |
287 | .class_name = "libmp3lame encoder", | |
288 | .item_name = av_default_item_name, | |
289 | .option = options, | |
290 | .version = LIBAVUTIL_VERSION_INT, | |
291 | }; | |
292 | ||
293 | static const AVCodecDefault libmp3lame_defaults[] = { | |
294 | { "b", "0" }, | |
295 | { NULL }, | |
296 | }; | |
297 | ||
298 | static const int libmp3lame_sample_rates[] = { | |
299 | 44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0 | |
300 | }; | |
301 | ||
302 | AVCodec ff_libmp3lame_encoder = { | |
303 | .name = "libmp3lame", | |
304 | .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"), | |
305 | .type = AVMEDIA_TYPE_AUDIO, | |
306 | .id = AV_CODEC_ID_MP3, | |
307 | .priv_data_size = sizeof(LAMEContext), | |
308 | .init = mp3lame_encode_init, | |
309 | .encode2 = mp3lame_encode_frame, | |
310 | .close = mp3lame_encode_close, | |
311 | .capabilities = CODEC_CAP_DELAY | CODEC_CAP_SMALL_LAST_FRAME, | |
312 | .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32P, | |
313 | AV_SAMPLE_FMT_FLTP, | |
314 | AV_SAMPLE_FMT_S16P, | |
315 | AV_SAMPLE_FMT_NONE }, | |
316 | .supported_samplerates = libmp3lame_sample_rates, | |
317 | .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO, | |
318 | AV_CH_LAYOUT_STEREO, | |
319 | 0 }, | |
320 | .priv_class = &libmp3lame_class, | |
321 | .defaults = libmp3lame_defaults, | |
322 | }; |