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libmp3lame: renaming, rearrangement, alignment, and comments
This commit is contained in:
Justin Ruggles
parent
232e16dd02
commit
e232225276
@ -24,6 +24,8 @@
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* Interface to libmp3lame for mp3 encoding.
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*/
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#include <lame/lame.h>
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#include "libavutil/intreadwrite.h"
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#include "libavutil/log.h"
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#include "libavutil/opt.h"
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@ -31,21 +33,21 @@
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#include "internal.h"
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#include "mpegaudio.h"
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#include "mpegaudiodecheader.h"
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#include <lame/lame.h>
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#define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4)
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typedef struct Mp3AudioContext {
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typedef struct LAMEContext {
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AVClass *class;
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lame_global_flags *gfp;
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uint8_t buffer[BUFFER_SIZE];
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int buffer_index;
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int reservoir;
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} Mp3AudioContext;
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} LAMEContext;
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static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
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static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
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{
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Mp3AudioContext *s = avctx->priv_data;
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LAMEContext *s = avctx->priv_data;
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av_freep(&avctx->coded_frame);
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@ -53,25 +55,34 @@ static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
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return 0;
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}
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static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
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static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
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{
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Mp3AudioContext *s = avctx->priv_data;
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LAMEContext *s = avctx->priv_data;
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int ret;
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if (avctx->channels > 2)
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return AVERROR(EINVAL);
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/* initialize LAME and get defaults */
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if ((s->gfp = lame_init()) == NULL)
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return AVERROR(ENOMEM);
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lame_set_in_samplerate(s->gfp, avctx->sample_rate);
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lame_set_out_samplerate(s->gfp, avctx->sample_rate);
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lame_set_num_channels(s->gfp, avctx->channels);
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if (avctx->compression_level == FF_COMPRESSION_DEFAULT) {
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lame_set_quality(s->gfp, 5);
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} else {
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lame_set_quality(s->gfp, avctx->compression_level);
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/* channels */
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if (avctx->channels > 2) {
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ret = AVERROR(EINVAL);
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goto error;
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}
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lame_set_num_channels(s->gfp, avctx->channels);
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lame_set_mode(s->gfp, avctx->channels > 1 ? JOINT_STEREO : MONO);
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/* sample rate */
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lame_set_in_samplerate (s->gfp, avctx->sample_rate);
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lame_set_out_samplerate(s->gfp, avctx->sample_rate);
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/* algorithmic quality */
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if (avctx->compression_level == FF_COMPRESSION_DEFAULT)
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lame_set_quality(s->gfp, 5);
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else
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lame_set_quality(s->gfp, avctx->compression_level);
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/* rate control */
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if (avctx->flags & CODEC_FLAG_QSCALE) {
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lame_set_VBR(s->gfp, vbr_default);
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lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
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@ -79,15 +90,21 @@ static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
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if (avctx->bit_rate)
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lame_set_brate(s->gfp, avctx->bit_rate / 1000);
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}
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/* do not get a Xing VBR header frame from LAME */
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lame_set_bWriteVbrTag(s->gfp,0);
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/* bit reservoir usage */
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lame_set_disable_reservoir(s->gfp, !s->reservoir);
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/* set specified parameters */
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if (lame_init_params(s->gfp) < 0) {
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ret = -1;
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goto error;
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}
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avctx->frame_size = lame_get_framesize(s->gfp);
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avctx->coded_frame = avcodec_alloc_frame();
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avctx->frame_size = lame_get_framesize(s->gfp);
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avctx->coded_frame = avcodec_alloc_frame();
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if (!avctx->coded_frame) {
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ret = AVERROR(ENOMEM);
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goto error;
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@ -95,18 +112,14 @@ static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
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return 0;
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error:
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MP3lame_encode_close(avctx);
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mp3lame_encode_close(avctx);
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return ret;
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}
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static const int sSampleRates[] = {
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44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
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};
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static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
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static int mp3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
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int buf_size, void *data)
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{
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Mp3AudioContext *s = avctx->priv_data;
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LAMEContext *s = avctx->priv_data;
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MPADecodeHeader hdr;
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int len;
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int lame_result;
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@ -127,7 +140,6 @@ static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
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lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
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BUFFER_SIZE - s->buffer_index);
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}
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if (lame_result < 0) {
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if (lame_result == -1) {
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av_log(avctx, AV_LOG_ERROR,
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@ -136,12 +148,13 @@ static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
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}
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return -1;
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}
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s->buffer_index += lame_result;
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/* Move 1 frame from the LAME buffer to the output packet, if available.
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We have to parse the first frame header in the output buffer to
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determine the frame size. */
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if (s->buffer_index < 4)
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return 0;
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if (avpriv_mpegaudio_decode_header(&hdr, AV_RB32(s->buffer))) {
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av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
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return -1;
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@ -152,14 +165,13 @@ static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
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if (len <= s->buffer_index) {
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memcpy(frame, s->buffer, len);
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s->buffer_index -= len;
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memmove(s->buffer, s->buffer + len, s->buffer_index);
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return len;
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} else
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return 0;
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}
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#define OFFSET(x) offsetof(Mp3AudioContext, x)
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#define OFFSET(x) offsetof(LAMEContext, x)
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#define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
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static const AVOption options[] = {
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{ "reservoir", "Use bit reservoir.", OFFSET(reservoir), AV_OPT_TYPE_INT, { 1 }, 0, 1, AE },
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@ -178,18 +190,22 @@ static const AVCodecDefault libmp3lame_defaults[] = {
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{ NULL },
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};
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static const int libmp3lame_sample_rates[] = {
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44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
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};
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AVCodec ff_libmp3lame_encoder = {
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.name = "libmp3lame",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = CODEC_ID_MP3,
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.priv_data_size = sizeof(Mp3AudioContext),
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.init = MP3lame_encode_init,
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.encode = MP3lame_encode_frame,
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.close = MP3lame_encode_close,
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.priv_data_size = sizeof(LAMEContext),
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.init = mp3lame_encode_init,
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.encode = mp3lame_encode_frame,
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.close = mp3lame_encode_close,
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.capabilities = CODEC_CAP_DELAY,
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_NONE },
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.supported_samplerates = sSampleRates,
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.supported_samplerates = libmp3lame_sample_rates,
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.long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
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.priv_class = &libmp3lame_class,
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.defaults = libmp3lame_defaults,
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