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Diffstat (limited to 'ffmpeg/libavcodec/libmp3lame.c')
-rw-r--r--ffmpeg/libavcodec/libmp3lame.c310
1 files changed, 0 insertions, 310 deletions
diff --git a/ffmpeg/libavcodec/libmp3lame.c b/ffmpeg/libavcodec/libmp3lame.c
deleted file mode 100644
index e6cb64b..0000000
--- a/ffmpeg/libavcodec/libmp3lame.c
+++ /dev/null
@@ -1,310 +0,0 @@
-/*
- * 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;
- 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;
-
- av_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);
-
- 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()) == NULL)
- 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, 5);
- else
- lame_set_quality(s->gfp, avctx->compression_level);
-
- /* rate control */
- if (avctx->flags & 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;
- }
-
- /* 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);
-
- /* 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;
- }
- }
- }
-
- ret = realloc_buffer(s);
- if (ret < 0)
- goto error;
-
- avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
-
- 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;
- int lame_result;
-
- 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,
- 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;
- 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) {
- if ((ret = ff_alloc_packet2(avctx, avpkt, len)) < 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);
-
- 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_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 },
-};
-
-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 = CODEC_CAP_DELAY | 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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