[FFmpeg-cvslog] libmp3lame: renaming, rearrangement, alignment, and comments
Justin Ruggles
git at videolan.org
Tue Feb 21 05:29:40 CET 2012
ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Fri Feb 17 00:04:54 2012 -0500| [e2322252764daad55dfe977dc3dba3e4e5ab67e1] | committer: Justin Ruggles
libmp3lame: renaming, rearrangement, alignment, and comments
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=e2322252764daad55dfe977dc3dba3e4e5ab67e1
---
libavcodec/libmp3lame.c | 86 ++++++++++++++++++++++++++++-------------------
1 files changed, 51 insertions(+), 35 deletions(-)
diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c
index 79384b8..365b639 100644
--- a/libavcodec/libmp3lame.c
+++ b/libavcodec/libmp3lame.c
@@ -24,6 +24,8 @@
* Interface to libmp3lame for mp3 encoding.
*/
+#include <lame/lame.h>
+
#include "libavutil/intreadwrite.h"
#include "libavutil/log.h"
#include "libavutil/opt.h"
@@ -31,21 +33,21 @@
#include "internal.h"
#include "mpegaudio.h"
#include "mpegaudiodecheader.h"
-#include <lame/lame.h>
#define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4)
-typedef struct Mp3AudioContext {
+
+typedef struct LAMEContext {
AVClass *class;
lame_global_flags *gfp;
uint8_t buffer[BUFFER_SIZE];
int buffer_index;
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);
@@ -53,25 +55,34 @@ static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
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;
- if (avctx->channels > 2)
- return AVERROR(EINVAL);
-
+ /* initialize LAME and get defaults */
if ((s->gfp = lame_init()) == NULL)
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);
- 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);
- } else {
+ else
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) {
lame_set_VBR(s->gfp, vbr_default);
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)
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;
}
- avctx->frame_size = lame_get_framesize(s->gfp);
- avctx->coded_frame = avcodec_alloc_frame();
+ avctx->frame_size = lame_get_framesize(s->gfp);
+ avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
@@ -95,18 +112,14 @@ static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
return 0;
error:
- MP3lame_encode_close(avctx);
+ mp3lame_encode_close(avctx);
return ret;
}
-static const int sSampleRates[] = {
- 44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
-};
-
-static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
+static int mp3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
int buf_size, void *data)
{
- Mp3AudioContext *s = avctx->priv_data;
+ LAMEContext *s = avctx->priv_data;
MPADecodeHeader hdr;
int len;
int lame_result;
@@ -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,
BUFFER_SIZE - s->buffer_index);
}
-
if (lame_result < 0) {
if (lame_result == -1) {
av_log(avctx, AV_LOG_ERROR,
@@ -136,12 +148,13 @@ static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
}
return -1;
}
-
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)
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;
@@ -152,14 +165,13 @@ static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
if (len <= s->buffer_index) {
memcpy(frame, s->buffer, len);
s->buffer_index -= len;
-
memmove(s->buffer, s->buffer + len, s->buffer_index);
return len;
} else
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
static const AVOption options[] = {
{ "reservoir", "Use bit reservoir.", OFFSET(reservoir), AV_OPT_TYPE_INT, { 1 }, 0, 1, AE },
@@ -178,18 +190,22 @@ static const AVCodecDefault libmp3lame_defaults[] = {
{ 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 = CODEC_ID_MP3,
- .priv_data_size = sizeof(Mp3AudioContext),
- .init = MP3lame_encode_init,
- .encode = MP3lame_encode_frame,
- .close = MP3lame_encode_close,
+ .priv_data_size = sizeof(LAMEContext),
+ .init = mp3lame_encode_init,
+ .encode = mp3lame_encode_frame,
+ .close = mp3lame_encode_close,
.capabilities = CODEC_CAP_DELAY,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
- .supported_samplerates = sSampleRates,
+ .supported_samplerates = libmp3lame_sample_rates,
.long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
.priv_class = &libmp3lame_class,
.defaults = libmp3lame_defaults,
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