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authorDiego Biurrun <diego@biurrun.de>2007-06-06 00:14:18 +0000
committerDiego Biurrun <diego@biurrun.de>2007-06-06 00:14:18 +0000
commit6f1af73557824b1872732b7823e881a14f870077 (patch)
tree3d22e9a9585b538bd4295898f1e2f5a89d36619b /libavcodec/libmp3lame.c
parent6f74b71ef0f5457b8cc736cb8f828ec1b75a217c (diff)
downloadffmpeg-6f1af73557824b1872732b7823e881a14f870077.tar.gz
Give all wrappers for external libraries names starting with lib.
Originally committed as revision 9226 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec/libmp3lame.c')
-rw-r--r--libavcodec/libmp3lame.c220
1 files changed, 220 insertions, 0 deletions
diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c
new file mode 100644
index 0000000000..9a5177af30
--- /dev/null
+++ b/libavcodec/libmp3lame.c
@@ -0,0 +1,220 @@
+/*
+ * 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 mp3lameaudio.c
+ * Interface to libmp3lame for mp3 encoding.
+ */
+
+#include "avcodec.h"
+#include "mpegaudio.h"
+#include <lame/lame.h>
+
+#define BUFFER_SIZE (2*MPA_FRAME_SIZE)
+typedef struct Mp3AudioContext {
+ lame_global_flags *gfp;
+ int stereo;
+ uint8_t buffer[BUFFER_SIZE];
+ int buffer_index;
+} Mp3AudioContext;
+
+static int MP3lame_encode_init(AVCodecContext *avctx)
+{
+ Mp3AudioContext *s = avctx->priv_data;
+
+ if (avctx->channels > 2)
+ return -1;
+
+ s->stereo = avctx->channels > 1 ? 1 : 0;
+
+ if ((s->gfp = lame_init()) == NULL)
+ goto err;
+ lame_set_in_samplerate(s->gfp, avctx->sample_rate);
+ lame_set_out_samplerate(s->gfp, avctx->sample_rate);
+ lame_set_num_channels(s->gfp, avctx->channels);
+ /* lame 3.91 dies on quality != 5 */
+ lame_set_quality(s->gfp, 5);
+ /* lame 3.91 doesn't work in mono */
+ lame_set_mode(s->gfp, JOINT_STEREO);
+ lame_set_brate(s->gfp, avctx->bit_rate/1000);
+ if(avctx->flags & CODEC_FLAG_QSCALE) {
+ lame_set_brate(s->gfp, 0);
+ lame_set_VBR(s->gfp, vbr_default);
+ lame_set_VBR_q(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
+ }
+ lame_set_bWriteVbrTag(s->gfp,0);
+ if (lame_init_params(s->gfp) < 0)
+ goto err_close;
+
+ avctx->frame_size = lame_get_framesize(s->gfp);
+
+ avctx->coded_frame= avcodec_alloc_frame();
+ avctx->coded_frame->key_frame= 1;
+
+ return 0;
+
+err_close:
+ lame_close(s->gfp);
+err:
+ return -1;
+}
+
+static const int sSampleRates[3] = {
+ 44100, 48000, 32000
+};
+
+static const int sBitRates[2][3][15] = {
+ { { 0, 32, 64, 96,128,160,192,224,256,288,320,352,384,416,448},
+ { 0, 32, 48, 56, 64, 80, 96,112,128,160,192,224,256,320,384},
+ { 0, 32, 40, 48, 56, 64, 80, 96,112,128,160,192,224,256,320}
+ },
+ { { 0, 32, 48, 56, 64, 80, 96,112,128,144,160,176,192,224,256},
+ { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160},
+ { 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96,112,128,144,160}
+ },
+};
+
+static const int sSamplesPerFrame[2][3] =
+{
+ { 384, 1152, 1152 },
+ { 384, 1152, 576 }
+};
+
+static const int sBitsPerSlot[3] = {
+ 32,
+ 8,
+ 8
+};
+
+static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
+{
+ uint32_t header = AV_RB32(data);
+ int layerID = 3 - ((header >> 17) & 0x03);
+ int bitRateID = ((header >> 12) & 0x0f);
+ int sampleRateID = ((header >> 10) & 0x03);
+ int bitsPerSlot = sBitsPerSlot[layerID];
+ int isPadded = ((header >> 9) & 0x01);
+ static int const mode_tab[4]= {2,3,1,0};
+ int mode= mode_tab[(header >> 19) & 0x03];
+ int mpeg_id= mode>0;
+ int temp0, temp1, bitRate;
+
+ if ( (( header >> 21 ) & 0x7ff) != 0x7ff || mode == 3 || layerID==3 || sampleRateID==3) {
+ return -1;
+ }
+
+ if(!samplesPerFrame) samplesPerFrame= &temp0;
+ if(!sampleRate ) sampleRate = &temp1;
+
+// *isMono = ((header >> 6) & 0x03) == 0x03;
+
+ *sampleRate = sSampleRates[sampleRateID]>>mode;
+ bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
+ *samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID];
+//av_log(NULL, AV_LOG_DEBUG, "sr:%d br:%d spf:%d l:%d m:%d\n", *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
+
+ return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
+}
+
+static int MP3lame_encode_frame(AVCodecContext *avctx,
+ unsigned char *frame, int buf_size, void *data)
+{
+ Mp3AudioContext *s = avctx->priv_data;
+ int len;
+ int lame_result;
+
+ /* lame 3.91 dies on '1-channel interleaved' data */
+
+ if(data){
+ if (s->stereo) {
+ lame_result = lame_encode_buffer_interleaved(
+ s->gfp,
+ data,
+ avctx->frame_size,
+ s->buffer + s->buffer_index,
+ BUFFER_SIZE - s->buffer_index
+ );
+ } else {
+ lame_result = lame_encode_buffer(
+ s->gfp,
+ data,
+ data,
+ avctx->frame_size,
+ s->buffer + s->buffer_index,
+ BUFFER_SIZE - s->buffer_index
+ );
+ }
+ }else{
+ lame_result= lame_encode_flush(
+ s->gfp,
+ s->buffer + s->buffer_index,
+ BUFFER_SIZE - s->buffer_index
+ );
+ }
+
+ if(lame_result==-1) {
+ /* output buffer too small */
+ av_log(avctx, AV_LOG_ERROR, "lame: output buffer too small (buffer index: %d, free bytes: %d)\n", s->buffer_index, BUFFER_SIZE - s->buffer_index);
+ return 0;
+ }
+
+ s->buffer_index += lame_result;
+
+ if(s->buffer_index<4)
+ return 0;
+
+ len= mp3len(s->buffer, NULL, NULL);
+//av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index);
+ if(len <= s->buffer_index){
+ memcpy(frame, s->buffer, len);
+ s->buffer_index -= len;
+
+ memmove(s->buffer, s->buffer+len, s->buffer_index);
+ //FIXME fix the audio codec API, so we dont need the memcpy()
+/*for(i=0; i<len; i++){
+ av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
+}*/
+ return len;
+ }else
+ return 0;
+}
+
+static int MP3lame_encode_close(AVCodecContext *avctx)
+{
+ Mp3AudioContext *s = avctx->priv_data;
+
+ av_freep(&avctx->coded_frame);
+
+ lame_close(s->gfp);
+ return 0;
+}
+
+
+AVCodec mp3lame_encoder = {
+ "mp3",
+ CODEC_TYPE_AUDIO,
+ CODEC_ID_MP3,
+ sizeof(Mp3AudioContext),
+ MP3lame_encode_init,
+ MP3lame_encode_frame,
+ MP3lame_encode_close,
+ .capabilities= CODEC_CAP_DELAY,
+};