27 #include <lame/lame.h> 41 #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. 46 lame_global_flags *
gfp;
99 if (!(s->
gfp = lame_init()))
116 lame_set_VBR(s->
gfp, vbr_default);
121 lame_set_VBR(s->
gfp, vbr_abr);
122 lame_set_VBR_mean_bitrate_kbps(s->
gfp, avctx->
bit_rate / 1000);
130 lame_set_lowpassfreq(s->
gfp, avctx->
cutoff);
133 lame_set_bWriteVbrTag(s->
gfp,0);
139 if (lame_init_params(s->
gfp) < 0) {
153 for (ch = 0; ch < avctx->
channels; ch++) {
180 #define ENCODE_BUFFER(func, buf_type, buf_name) do { \ 181 lame_result = func(s->gfp, \ 182 (const buf_type *)buf_name[0], \ 183 (const buf_type *)buf_name[1], frame->nb_samples, \ 184 s->buffer + s->buffer_index, \ 185 s->buffer_size - s->buffer_index); \ 193 int len,
ret, ch, discard_padding;
210 for (ch = 0; ch < avctx->
channels; ch++) {
212 (
const float *)frame->
data[ch],
227 if (lame_result < 0) {
228 if (lame_result == -1) {
230 "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
263 len = hdr.frame_size;
296 AV_WL32(side_data + 4, discard_padding);
305 #define OFFSET(x) offsetof(LAMEContext, x) 306 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM 327 44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
331 .
name =
"libmp3lame",
350 .wrapper_name =
"libmp3lame",
static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int64_t *duration)
Remove frame(s) from the queue.
static const AVClass libmp3lame_class
#define FF_COMPRESSION_DEFAULT
This structure describes decoded (raw) audio or video data.
int64_t bit_rate
the average bitrate
#define LIBAVUTIL_VERSION_INT
static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
static av_cold int init(AVCodecContext *avctx)
const char * av_default_item_name(void *ptr)
Return the context name.
static const int libmp3lame_sample_rates[]
AVCodec ff_libmp3lame_encoder
#define AV_CH_LAYOUT_STEREO
static void error(const char *err)
static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
#define AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
enum AVSampleFormat sample_fmt
audio sample format
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
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 AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
int initial_padding
Audio only.
preferred ID for decoding MPEG audio layer 1, 2 or 3
int flags
AV_CODEC_FLAG_*.
const char * name
Name of the codec implementation.
static const AVOption options[]
static const AVCodecDefault libmp3lame_defaults[]
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
static int realloc_buffer(LAMEContext *s)
audio channel layout utility functions
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
#define AV_CODEC_FLAG_QSCALE
Use fixed qscale.
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
int frame_size
Number of samples per channel in an audio frame.
int av_reallocp(void *ptr, size_t size)
Allocate, reallocate, or free a block of memory through a pointer to a pointer.
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
int sample_rate
samples per second
int linesize[AV_NUM_DATA_POINTERS]
For video, size in bytes of each picture line.
main external API structure.
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
Describe the class of an AVClass context structure.
Recommmends skipping the specified number of samples.
void(* vector_fmul_scalar)(float *dst, const float *src, float mul, int len)
Multiply a vector of floats by a scalar float.
int global_quality
Global quality for codecs which cannot change it per frame.
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
common internal api header.
common internal and external API header
mpeg audio declarations for both encoder and decoder.
#define ENCODE_BUFFER(func, buf_type, buf_name)
int cutoff
Audio cutoff bandwidth (0 means "automatic")
int channels
number of audio channels
#define FF_QP2LAMBDA
factor to convert from H.263 QP to lambda
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
static enum AVSampleFormat sample_fmts[]
#define av_malloc_array(a, b)
uint8_t * av_packet_new_side_data(AVPacket *pkt, enum AVPacketSideDataType type, int size)
Allocate new information of a packet.
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
#define AV_CH_LAYOUT_MONO
This structure stores compressed data.
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...