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Diffstat (limited to 'ffmpeg1/libavcodec/libmp3lame.c')
-rw-r--r--ffmpeg1/libavcodec/libmp3lame.c305
1 files changed, 305 insertions, 0 deletions
diff --git a/ffmpeg1/libavcodec/libmp3lame.c b/ffmpeg1/libavcodec/libmp3lame.c
new file mode 100644
index 0000000..94e3582
--- /dev/null
+++ b/ffmpeg1/libavcodec/libmp3lame.c
@@ -0,0 +1,305 @@
+/*
+ * 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;
+ 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) {
+ uint8_t *tmp;
+ int new_size = s->buffer_index + 2 * BUFFER_SIZE;
+
+ av_dlog(s->avctx, "resizing output buffer: %d -> %d\n", s->buffer_size,
+ new_size);
+ tmp = av_realloc(s->buffer, new_size);
+ if (!tmp) {
+ av_freep(&s->buffer);
+ s->buffer_size = s->buffer_index = 0;
+ return AVERROR(ENOMEM);
+ }
+ s->buffer = tmp;
+ 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 ? 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
+ lame_set_quality(s->gfp, avctx->compression_level);
+
+ /* rate control */
+ if (avctx->flags & CODEC_FLAG_QSCALE) {
+ 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;
+ }
+
+ /* 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 },
+ { 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",
+ .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 },
+ .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
+ .priv_class = &libmp3lame_class,
+ .defaults = libmp3lame_defaults,
+};
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