Commit e2322252 authored by Justin Ruggles's avatar Justin Ruggles

libmp3lame: renaming, rearrangement, alignment, and comments

parent 232e16dd
...@@ -24,6 +24,8 @@ ...@@ -24,6 +24,8 @@
* Interface to libmp3lame for mp3 encoding. * Interface to libmp3lame for mp3 encoding.
*/ */
#include <lame/lame.h>
#include "libavutil/intreadwrite.h" #include "libavutil/intreadwrite.h"
#include "libavutil/log.h" #include "libavutil/log.h"
#include "libavutil/opt.h" #include "libavutil/opt.h"
...@@ -31,21 +33,21 @@ ...@@ -31,21 +33,21 @@
#include "internal.h" #include "internal.h"
#include "mpegaudio.h" #include "mpegaudio.h"
#include "mpegaudiodecheader.h" #include "mpegaudiodecheader.h"
#include <lame/lame.h>
#define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4) #define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4)
typedef struct Mp3AudioContext {
typedef struct LAMEContext {
AVClass *class; AVClass *class;
lame_global_flags *gfp; lame_global_flags *gfp;
uint8_t buffer[BUFFER_SIZE]; uint8_t buffer[BUFFER_SIZE];
int buffer_index; int buffer_index;
int reservoir; int reservoir;
} Mp3AudioContext; } LAMEContext;
static av_cold int MP3lame_encode_close(AVCodecContext *avctx) static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
{ {
Mp3AudioContext *s = avctx->priv_data; LAMEContext *s = avctx->priv_data;
av_freep(&avctx->coded_frame); av_freep(&avctx->coded_frame);
...@@ -53,25 +55,34 @@ static av_cold int MP3lame_encode_close(AVCodecContext *avctx) ...@@ -53,25 +55,34 @@ static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
return 0; return 0;
} }
static av_cold int MP3lame_encode_init(AVCodecContext *avctx) static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
{ {
Mp3AudioContext *s = avctx->priv_data; LAMEContext *s = avctx->priv_data;
int ret; int ret;
if (avctx->channels > 2) /* initialize LAME and get defaults */
return AVERROR(EINVAL);
if ((s->gfp = lame_init()) == NULL) if ((s->gfp = lame_init()) == NULL)
return AVERROR(ENOMEM); return AVERROR(ENOMEM);
lame_set_in_samplerate(s->gfp, avctx->sample_rate);
lame_set_out_samplerate(s->gfp, avctx->sample_rate); /* channels */
if (avctx->channels > 2) {
ret = AVERROR(EINVAL);
goto error;
}
lame_set_num_channels(s->gfp, avctx->channels); lame_set_num_channels(s->gfp, avctx->channels);
if (avctx->compression_level == FF_COMPRESSION_DEFAULT) { 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); lame_set_quality(s->gfp, 5);
} else { else
lame_set_quality(s->gfp, avctx->compression_level); lame_set_quality(s->gfp, avctx->compression_level);
}
lame_set_mode(s->gfp, avctx->channels > 1 ? JOINT_STEREO : MONO); /* rate control */
if (avctx->flags & CODEC_FLAG_QSCALE) { if (avctx->flags & CODEC_FLAG_QSCALE) {
lame_set_VBR(s->gfp, vbr_default); lame_set_VBR(s->gfp, vbr_default);
lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA); lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
...@@ -79,15 +90,21 @@ static av_cold int MP3lame_encode_init(AVCodecContext *avctx) ...@@ -79,15 +90,21 @@ static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
if (avctx->bit_rate) if (avctx->bit_rate)
lame_set_brate(s->gfp, avctx->bit_rate / 1000); 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); lame_set_bWriteVbrTag(s->gfp,0);
/* bit reservoir usage */
lame_set_disable_reservoir(s->gfp, !s->reservoir); lame_set_disable_reservoir(s->gfp, !s->reservoir);
/* set specified parameters */
if (lame_init_params(s->gfp) < 0) { if (lame_init_params(s->gfp) < 0) {
ret = -1; ret = -1;
goto error; goto error;
} }
avctx->frame_size = lame_get_framesize(s->gfp); avctx->frame_size = lame_get_framesize(s->gfp);
avctx->coded_frame = avcodec_alloc_frame(); avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame) { if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM); ret = AVERROR(ENOMEM);
goto error; goto error;
...@@ -95,18 +112,14 @@ static av_cold int MP3lame_encode_init(AVCodecContext *avctx) ...@@ -95,18 +112,14 @@ static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
return 0; return 0;
error: error:
MP3lame_encode_close(avctx); mp3lame_encode_close(avctx);
return ret; return ret;
} }
static const int sSampleRates[] = { static int mp3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
};
static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
int buf_size, void *data) int buf_size, void *data)
{ {
Mp3AudioContext *s = avctx->priv_data; LAMEContext *s = avctx->priv_data;
MPADecodeHeader hdr; MPADecodeHeader hdr;
int len; int len;
int lame_result; int lame_result;
...@@ -127,7 +140,6 @@ static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame, ...@@ -127,7 +140,6 @@ static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index, lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index); BUFFER_SIZE - s->buffer_index);
} }
if (lame_result < 0) { if (lame_result < 0) {
if (lame_result == -1) { if (lame_result == -1) {
av_log(avctx, AV_LOG_ERROR, av_log(avctx, AV_LOG_ERROR,
...@@ -136,12 +148,13 @@ static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame, ...@@ -136,12 +148,13 @@ static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
} }
return -1; return -1;
} }
s->buffer_index += lame_result; s->buffer_index += lame_result;
/* 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) if (s->buffer_index < 4)
return 0; return 0;
if (avpriv_mpegaudio_decode_header(&hdr, AV_RB32(s->buffer))) { if (avpriv_mpegaudio_decode_header(&hdr, AV_RB32(s->buffer))) {
av_log(avctx, AV_LOG_ERROR, "free format output not supported\n"); av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
return -1; return -1;
...@@ -152,14 +165,13 @@ static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame, ...@@ -152,14 +165,13 @@ static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
if (len <= s->buffer_index) { if (len <= s->buffer_index) {
memcpy(frame, s->buffer, len); memcpy(frame, s->buffer, len);
s->buffer_index -= len; s->buffer_index -= len;
memmove(s->buffer, s->buffer + len, s->buffer_index); memmove(s->buffer, s->buffer + len, s->buffer_index);
return len; return len;
} else } else
return 0; return 0;
} }
#define OFFSET(x) offsetof(Mp3AudioContext, x) #define OFFSET(x) offsetof(LAMEContext, x)
#define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
static const AVOption options[] = { static const AVOption options[] = {
{ "reservoir", "Use bit reservoir.", OFFSET(reservoir), AV_OPT_TYPE_INT, { 1 }, 0, 1, AE }, { "reservoir", "Use bit reservoir.", OFFSET(reservoir), AV_OPT_TYPE_INT, { 1 }, 0, 1, AE },
...@@ -178,18 +190,22 @@ static const AVCodecDefault libmp3lame_defaults[] = { ...@@ -178,18 +190,22 @@ static const AVCodecDefault libmp3lame_defaults[] = {
{ NULL }, { NULL },
}; };
static const int libmp3lame_sample_rates[] = {
44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
};
AVCodec ff_libmp3lame_encoder = { AVCodec ff_libmp3lame_encoder = {
.name = "libmp3lame", .name = "libmp3lame",
.type = AVMEDIA_TYPE_AUDIO, .type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_MP3, .id = CODEC_ID_MP3,
.priv_data_size = sizeof(Mp3AudioContext), .priv_data_size = sizeof(LAMEContext),
.init = MP3lame_encode_init, .init = mp3lame_encode_init,
.encode = MP3lame_encode_frame, .encode = mp3lame_encode_frame,
.close = MP3lame_encode_close, .close = mp3lame_encode_close,
.capabilities = CODEC_CAP_DELAY, .capabilities = CODEC_CAP_DELAY,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16, .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE }, AV_SAMPLE_FMT_NONE },
.supported_samplerates = sSampleRates, .supported_samplerates = libmp3lame_sample_rates,
.long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"), .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
.priv_class = &libmp3lame_class, .priv_class = &libmp3lame_class,
.defaults = libmp3lame_defaults, .defaults = libmp3lame_defaults,
......
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