[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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