23 #include <opus_multistream.h>
25 #include "libavutil/opt.h"
53 0, 1, 1, 2, 2, 2, 2, 3
64 { 0, 4, 1, 2, 3, 5, 6 },
65 { 0, 6, 1, 2, 3, 4, 5, 7 },
76 { 0, 1, 5, 6, 2, 4, 3 },
77 { 0, 1, 6, 7, 4, 5, 2, 3 },
81 int coupled_stream_count,
88 bytestream_put_byte(&p, 1);
89 bytestream_put_byte(&p, channels);
90 bytestream_put_le16(&p, avctx->
delay);
92 bytestream_put_le16(&p, 0);
96 bytestream_put_byte(&p, channels <= 8 ? 1 : 255);
97 bytestream_put_byte(&p, stream_count);
98 bytestream_put_byte(&p, coupled_stream_count);
101 bytestream_put_byte(&p, 0);
112 "Quality-based encoding not supported, "
113 "please specify a bitrate and VBR setting.\n");
117 ret = opus_multistream_encoder_ctl(enc, OPUS_SET_BITRATE(avctx->
bit_rate));
118 if (ret != OPUS_OK) {
120 "Failed to set bitrate: %s\n", opus_strerror(ret));
124 ret = opus_multistream_encoder_ctl(enc,
128 "Unable to set complexity: %s\n", opus_strerror(ret));
130 ret = opus_multistream_encoder_ctl(enc, OPUS_SET_VBR(!!opts->
vbr));
133 "Unable to set VBR: %s\n", opus_strerror(ret));
135 ret = opus_multistream_encoder_ctl(enc,
136 OPUS_SET_VBR_CONSTRAINT(opts->
vbr == 2));
139 "Unable to set constrained VBR: %s\n", opus_strerror(ret));
141 ret = opus_multistream_encoder_ctl(enc,
145 "Unable to set expected packet loss percentage: %s\n",
149 ret = opus_multistream_encoder_ctl(enc,
153 "Unable to set maximum bandwidth: %s\n", opus_strerror(ret));
162 const uint8_t *channel_mapping;
165 int coupled_stream_count, header_size,
frame_size;
175 "Channel layout undefined for %d channels.\n", avctx->
channels);
180 32000 * coupled_stream_count;
182 "No bit rate set. Defaulting to %d bps.\n", avctx->
bit_rate);
187 "Please choose a value between 500 and %d.\n", avctx->
bit_rate,
193 switch (frame_size) {
198 "LPC mode cannot be used with a frame duration of less "
199 "than 10ms. Enabling restricted low-delay mode.\n"
200 "Use a longer frame duration if this is not what you want.\n");
213 "Frame duration must be exactly one of: 2.5, 5, 10, 20, 40 or 60.\n",
220 "Compression level must be in the range 0 to 10. "
221 "Defaulting to 10.\n");
246 "Invalid frequency cutoff: %d. Using default maximum bandwidth.\n"
247 "Cutoff frequency must be exactly one of: 4000, 6000, 8000, 12000 or 20000.\n",
255 coupled_stream_count,
258 if (ret != OPUS_OK) {
260 "Failed to create encoder: %s\n", opus_strerror(ret));
265 if (ret != OPUS_OK) {
287 ret = opus_multistream_encoder_ctl(enc, OPUS_GET_LOOKAHEAD(&avctx->
delay));
290 "Unable to get number of lookahead samples: %s\n",
303 opus_multistream_encoder_destroy(enc);
312 const int sample_size = avctx->
channels *
324 audio = frame->
data[0];
339 ret = opus_multistream_encode_float(opus->
enc, (
float *)audio,
343 ret = opus_multistream_encode(opus->
enc, (opus_int16 *)audio,
349 "Error encoding frame: %s\n", opus_strerror(ret));
360 if ((discard_padding < opus->opts.packet_size) != (avpkt->
duration > 0)) {
365 if (discard_padding > 0) {
369 if(side_data == NULL) {
374 AV_WL32(side_data + 4, discard_padding);
386 opus_multistream_encoder_destroy(opus->
enc);
396 #define OFFSET(x) offsetof(LibopusEncContext, opts.x)
397 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
399 {
"application",
"Intended application type",
OFFSET(application),
AV_OPT_TYPE_INT, { .i64 = OPUS_APPLICATION_AUDIO }, OPUS_APPLICATION_VOIP, OPUS_APPLICATION_RESTRICTED_LOWDELAY,
FLAGS,
"application" },
400 {
"voip",
"Favor improved speech intelligibility", 0,
AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_VOIP }, 0, 0,
FLAGS,
"application" },
401 {
"audio",
"Favor faithfulness to the input", 0,
AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_AUDIO }, 0, 0,
FLAGS,
"application" },
402 {
"lowdelay",
"Restrict to only the lowest delay modes", 0,
AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_RESTRICTED_LOWDELAY }, 0, 0,
FLAGS,
"application" },
403 {
"frame_duration",
"Duration of a frame in milliseconds",
OFFSET(frame_duration),
AV_OPT_TYPE_FLOAT, { .dbl = 10.0 }, 2.5, 60.0,
FLAGS },
421 {
"compression_level",
"10" },
426 48000, 24000, 16000, 12000, 8000, 0,
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size)
Check AVPacket size and/or allocate data.
void av_free_packet(AVPacket *pkt)
Free a packet.
This structure describes decoded (raw) audio or video data.
static int libopus_configure_encoder(AVCodecContext *avctx, OpusMSEncoder *enc, LibopusEncOpts *opts)
#define AV_LOG_WARNING
Something somehow does not look correct.
#define LIBAVUTIL_VERSION_INT
void av_shrink_packet(AVPacket *pkt, int size)
Reduce packet size, correctly zeroing padding.
static av_cold int init(AVCodecContext *avctx)
void av_log(void *avcl, int level, const char *fmt,...) av_printf_format(3
Send the specified message to the log if the level is less than or equal to the current av_log_level...
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
void av_freep(void *ptr)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc() and set the pointer ...
int ff_opus_error_to_averror(int err)
const char * av_default_item_name(void *ctx)
Return the context name.
enum AVSampleFormat sample_fmt
audio sample format
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
static const AVClass libopus_class
const char * name
Name of the codec implementation.
int duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
static const uint8_t frame_size[4]
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
void av_free(void *ptr)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc(). ...
#define CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
#define CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
static const uint8_t libavcodec_libopus_channel_map[8][8]
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
Libavcodec external API header.
uint8_t * av_packet_new_side_data(AVPacket *pkt, enum AVPacketSideDataType type, int size)
Allocate new information of a packet.
static const uint8_t opus_coupled_streams[8]
#define FF_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
int bit_rate
the average bitrate
void * av_malloc(size_t size) av_malloc_attrib 1(1)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
static const int libopus_sample_rates[]
int frame_size
Number of samples per channel in an audio frame.
AVSampleFormat
Audio Sample Formats.
static av_cold int libopus_encode_init(AVCodecContext *avctx)
int sample_rate
samples per second
main external API structure.
static int libopus_encode(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
static void close(AVCodecParserContext *s)
const uint64_t ff_vorbis_channel_layouts[9]
AVCodec ff_libopus_encoder
Describe the class of an AVClass context structure.
Recommmends skipping the specified number of samples.
static av_cold int libopus_encode_close(AVCodecContext *avctx)
int global_quality
Global quality for codecs which cannot change it per frame.
static const uint8_t opus_vorbis_channel_map[8][8]
common internal api header.
static const AVCodecDefault libopus_defaults[]
static av_always_inline void bytestream_put_buffer(uint8_t **b, const uint8_t *src, unsigned int size)
int cutoff
Audio cutoff bandwidth (0 means "automatic")
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int *duration)
Remove frame(s) from the queue.
int channels
number of audio channels
static const AVOption libopus_options[]
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
static enum AVSampleFormat sample_fmts[]
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
This structure stores compressed data.
int nb_samples
number of audio samples (per channel) described by this frame
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
static void libopus_write_header(AVCodecContext *avctx, int stream_count, int coupled_stream_count, const uint8_t *channel_mapping)
void * av_mallocz(size_t size) av_malloc_attrib 1(1)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...