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27 #include <lame/lame.h>
42 #define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4+1000) // FIXME: Buffer size to small? Adding 1000 to make up for it.
47 lame_global_flags *
gfp;
65 if (!
s->buffer ||
s->buffer_size -
s->buffer_index <
BUFFER_SIZE) {
68 ff_dlog(
s->avctx,
"resizing output buffer: %d -> %d\n",
s->buffer_size,
71 s->buffer_size =
s->buffer_index = 0;
74 s->buffer_size = new_size;
102 if (!(
s->gfp = lame_init()))
120 lame_set_VBR(
s->gfp, vbr_default);
125 lame_set_VBR(
s->gfp, vbr_abr);
126 lame_set_VBR_mean_bitrate_kbps(
s->gfp, avctx->
bit_rate / 1000);
128 lame_set_brate(
s->gfp, avctx->
bit_rate / 1000);
134 lame_set_lowpassfreq(
s->gfp, avctx->
cutoff);
137 lame_set_bWriteVbrTag(
s->gfp,0);
140 lame_set_disable_reservoir(
s->gfp, !
s->reservoir);
143 lame_set_copyright(
s->gfp,
s->copyright);
146 lame_set_original(
s->gfp,
s->original);
149 if (lame_init_params(
s->gfp) < 0) {
165 sizeof(*
s->samples_flt[ch]));
166 if (!
s->samples_flt[ch]) {
190 #define ENCODE_BUFFER(func, buf_type, buf_name) do { \
191 lame_result = func(s->gfp, \
192 (const buf_type *)buf_name[0], \
193 (const buf_type *)buf_name[1], frame->nb_samples, \
194 s->buffer + s->buffer_index, \
195 s->buffer_size - s->buffer_index); \
203 int len,
ret, ch, discard_padding;
221 s->fdsp->vector_fmul_scalar(
s->samples_flt[ch],
231 }
else if (!
s->afq.frame_alloc) {
234 lame_result = lame_encode_flush(
s->gfp,
s->buffer +
s->buffer_index,
235 s->buffer_size -
s->buffer_index);
237 if (lame_result < 0) {
238 if (lame_result == -1) {
240 "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
241 s->buffer_index,
s->buffer_size -
s->buffer_index);
245 s->buffer_index += lame_result;
261 if (
s->buffer_index < 4)
273 len = hdr.frame_size;
276 if (len <= s->buffer_index) {
279 memcpy(avpkt->
data,
s->buffer,
len);
280 s->buffer_index -=
len;
281 memmove(
s->buffer,
s->buffer +
len,
s->buffer_index);
293 if ((!
s->delay_sent && avctx->
initial_padding > 0) || discard_padding > 0) {
299 if (!
s->delay_sent) {
303 AV_WL32(side_data + 4, discard_padding);
311 #define OFFSET(x) offsetof(LAMEContext, x)
312 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
335 44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
339 .
p.
name =
"libmp3lame",
362 .p.wrapper_name =
"libmp3lame",
static void error(const char *err)
int frame_size
Number of samples per channel in an audio frame.
@ AV_SAMPLE_FMT_FLTP
float, planar
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int64_t *duration)
Remove frame(s) from the queue.
#define AV_CHANNEL_LAYOUT_STEREO
int sample_rate
samples per second
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
#define AV_CH_LAYOUT_MONO
#define AV_CODEC_FLAG_QSCALE
Use fixed qscale.
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
This structure describes decoded (raw) audio or video data.
@ AV_SAMPLE_FMT_S32P
signed 32 bits, planar
#define FF_CODEC_CAP_NOT_INIT_THREADSAFE
The codec is not known to be init-threadsafe (i.e.
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
int nb_channels
Number of channels in this layout.
#define FF_COMPRESSION_DEFAULT
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
AVCodec p
The public AVCodec.
AVChannelLayout ch_layout
Audio channel layout.
int initial_padding
Audio only.
int flags
AV_CODEC_FLAG_*.
@ AV_CODEC_ID_MP3
preferred ID for decoding MPEG audio layer 1, 2 or 3
#define FF_CODEC_ENCODE_CB(func)
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
#define AV_CH_LAYOUT_STEREO
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static const FFCodecDefault libmp3lame_defaults[]
static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
int global_quality
Global quality for codecs which cannot change it per frame.
int(* init)(AVBSFContext *ctx)
#define CODEC_OLD_CHANNEL_LAYOUTS(...)
static int realloc_buffer(LAMEContext *s)
#define CODEC_LONG_NAME(str)
#define LIBAVUTIL_VERSION_INT
Describe the class of an AVClass context structure.
int64_t bit_rate
the average bitrate
const char * av_default_item_name(void *ptr)
Return the context name.
static const AVClass libmp3lame_class
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
An AVChannelLayout holds information about the channel layout of audio data.
enum AVSampleFormat sample_fmt
audio sample format
int av_reallocp(void *ptr, size_t size)
Allocate, reallocate, or free a block of memory through a pointer to a pointer.
#define ENCODE_BUFFER(func, buf_type, buf_name)
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_RB32
#define AVERROR_EXTERNAL
Generic error in an external library.
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
int nb_samples
number of audio samples (per channel) described by this frame
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
#define av_malloc_array(a, b)
int cutoff
Audio cutoff bandwidth (0 means "automatic")
AVSampleFormat
Audio sample formats.
const char * name
Name of the codec implementation.
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
main external API structure.
const FFCodec ff_libmp3lame_encoder
uint8_t * av_packet_new_side_data(AVPacket *pkt, enum AVPacketSideDataType type, size_t size)
Allocate new information of a packet.
int ff_get_encode_buffer(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int flags)
Get a buffer for a packet.
static const AVOption options[]
static const int libmp3lame_sample_rates[]
@ AV_PKT_DATA_SKIP_SAMPLES
Recommmends skipping the specified number of samples.
#define AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
#define AV_CHANNEL_LAYOUT_MONO
This structure stores compressed data.
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
int linesize[AV_NUM_DATA_POINTERS]
For video, a positive or negative value, which is typically indicating the size in bytes of each pict...
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
#define FF_QP2LAMBDA
factor to convert from H.263 QP to lambda
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.