23 #include <opus_multistream.h>
55 0, 1, 1, 2, 2, 2, 2, 3
66 { 0, 4, 1, 2, 3, 5, 6 },
67 { 0, 6, 1, 2, 3, 4, 5, 7 },
78 { 0, 1, 5, 6, 2, 4, 3 },
79 { 0, 1, 6, 7, 4, 5, 2, 3 },
83 int coupled_stream_count,
91 bytestream_put_byte(&p, 1);
92 bytestream_put_byte(&p, channels);
95 bytestream_put_le16(&p, 0);
98 bytestream_put_byte(&p, mapping_family);
99 if (mapping_family != 0) {
100 bytestream_put_byte(&p, stream_count);
101 bytestream_put_byte(&p, coupled_stream_count);
113 "Quality-based encoding not supported, "
114 "please specify a bitrate and VBR setting.\n");
118 ret = opus_multistream_encoder_ctl(enc, OPUS_SET_BITRATE(avctx->
bit_rate));
119 if (ret != OPUS_OK) {
121 "Failed to set bitrate: %s\n", opus_strerror(ret));
125 ret = opus_multistream_encoder_ctl(enc,
129 "Unable to set complexity: %s\n", opus_strerror(ret));
131 ret = opus_multistream_encoder_ctl(enc, OPUS_SET_VBR(!!opts->
vbr));
134 "Unable to set VBR: %s\n", opus_strerror(ret));
136 ret = opus_multistream_encoder_ctl(enc,
137 OPUS_SET_VBR_CONSTRAINT(opts->
vbr == 2));
140 "Unable to set constrained VBR: %s\n", opus_strerror(ret));
142 ret = opus_multistream_encoder_ctl(enc,
146 "Unable to set expected packet loss percentage: %s\n",
150 ret = opus_multistream_encoder_ctl(enc,
154 "Unable to set maximum bandwidth: %s\n", opus_strerror(ret));
162 if (avctx->
channels > max_channels) {
176 "No channel layout specified. Opus encoder will use Vorbis "
177 "channel layout for %d channels.\n", avctx->
channels);
183 "Invalid channel layout %s for specified mapping family %d.\n",
184 name, mapping_family);
195 const uint8_t ** channel_map_result)
200 switch (mapping_family) {
228 "Unknown channel mapping family %d. Output channel layout may be invalid.\n",
233 *channel_map_result = channel_map;
241 uint8_t libopus_channel_mapping[255];
244 int coupled_stream_count, header_size,
frame_size;
248 switch (frame_size) {
253 "LPC mode cannot be used with a frame duration of less "
254 "than 10ms. Enabling restricted low-delay mode.\n"
255 "Use a longer frame duration if this is not what you want.\n");
268 "Frame duration must be exactly one of: 2.5, 5, 10, 20, 40 or 60.\n",
275 "Compression level must be in the range 0 to 10. "
276 "Defaulting to 10.\n");
301 "Invalid frequency cutoff: %d. Using default maximum bandwidth.\n"
302 "Cutoff frequency must be exactly one of: 4000, 6000, 8000, 12000 or 20000.\n",
320 mapping_family = avctx->
channels > 2 ? 1 : 0;
323 memcpy(libopus_channel_mapping,
325 avctx->
channels *
sizeof(*libopus_channel_mapping));
327 enc = opus_multistream_encoder_create(
329 coupled_stream_count,
337 enc = opus_multistream_surround_encoder_create(
339 &opus->
stream_count, &coupled_stream_count, libopus_channel_mapping,
343 if (ret != OPUS_OK) {
345 "Failed to create encoder: %s\n", opus_strerror(ret));
352 32000 * coupled_stream_count;
354 "No bit rate set. Defaulting to %"PRId64
" bps.\n", avctx->
bit_rate);
359 "Please choose a value between 500 and %d.\n", avctx->
bit_rate,
366 if (ret != OPUS_OK) {
372 header_size = 19 + (mapping_family == 0 ? 0 : 2 + avctx->
channels);
389 ret = opus_multistream_encoder_ctl(enc, OPUS_GET_LOOKAHEAD(&avctx->
initial_padding));
392 "Unable to get number of lookahead samples: %s\n",
396 mapping_family, libopus_channel_mapping);
405 opus_multistream_encoder_destroy(enc);
412 int nb_channels,
int nb_samples,
int bytes_per_sample) {
414 for (sample = 0; sample < nb_samples; ++
sample) {
416 const size_t src_pos = bytes_per_sample * (nb_channels * sample +
channel);
417 const size_t dst_pos = bytes_per_sample * (nb_channels * sample + channel_map[
channel]);
419 memcpy(&dst[dst_pos], &src[src_pos], bytes_per_sample);
429 const int sample_size = avctx->
channels * bytes_per_sample;
447 audio = frame->
data[0];
462 ret = opus_multistream_encode_float(opus->
enc, (
float *)audio,
466 ret = opus_multistream_encode(opus->
enc, (opus_int16 *)audio,
472 "Error encoding frame: %s\n", opus_strerror(ret));
483 if ((discard_padding < opus->
opts.packet_size) != (avpkt->
duration > 0)) {
488 if (discard_padding > 0) {
497 AV_WL32(side_data + 4, discard_padding);
509 opus_multistream_encoder_destroy(opus->
enc);
519 #define OFFSET(x) offsetof(LibopusEncContext, opts.x)
520 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
522 {
"application",
"Intended application type",
OFFSET(application),
AV_OPT_TYPE_INT, { .i64 = OPUS_APPLICATION_AUDIO }, OPUS_APPLICATION_VOIP, OPUS_APPLICATION_RESTRICTED_LOWDELAY,
FLAGS,
"application" },
523 {
"voip",
"Favor improved speech intelligibility", 0,
AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_VOIP }, 0, 0,
FLAGS,
"application" },
524 {
"audio",
"Favor faithfulness to the input", 0,
AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_AUDIO }, 0, 0,
FLAGS,
"application" },
525 {
"lowdelay",
"Restrict to only the lowest delay modes", 0,
AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_RESTRICTED_LOWDELAY }, 0, 0,
FLAGS,
"application" },
526 {
"frame_duration",
"Duration of a frame in milliseconds",
OFFSET(frame_duration),
AV_OPT_TYPE_FLOAT, { .dbl = 20.0 }, 2.5, 60.0,
FLAGS },
532 {
"mapping_family",
"Channel Mapping Family",
OFFSET(mapping_family),
AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 255,
FLAGS,
"mapping_family" },
545 {
"compression_level",
"10" },
550 48000, 24000, 16000, 12000, 8000, 0,
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int64_t *duration)
Remove frame(s) from the queue.
This structure describes decoded (raw) audio or video data.
static int libopus_configure_encoder(AVCodecContext *avctx, OpusMSEncoder *enc, LibopusEncOpts *opts)
static int libopus_check_vorbis_layout(AVCodecContext *avctx, int mapping_family)
#define AV_LOG_WARNING
Something somehow does not look correct.
int64_t bit_rate
the average bitrate
#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)
static int libopus_validate_layout_and_get_channel_map(AVCodecContext *avctx, int mapping_family, const uint8_t **channel_map_result)
static void libopus_write_header(AVCodecContext *avctx, int stream_count, int coupled_stream_count, int mapping_family, const uint8_t *channel_mapping)
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_opus_error_to_averror(int err)
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
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
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.
static const AVClass libopus_class
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
const uint8_t * encoder_channel_map
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static int libopus_check_max_channels(AVCodecContext *avctx, int max_channels)
#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 initial_padding
Audio only.
const char * name
Name of the codec implementation.
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
static void * av_mallocz_array(size_t nmemb, size_t size)
uint64_t channel_layout
Audio channel layout.
static const uint8_t opus_coupled_streams[8]
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
static const int libopus_sample_rates[]
#define FF_ARRAY_ELEMS(a)
void av_get_channel_layout_string(char *buf, int buf_size, int nb_channels, uint64_t channel_layout)
Return a description of a channel layout.
static void libopus_copy_samples_with_channel_map(uint8_t *dst, const uint8_t *src, const uint8_t *channel_map, int nb_channels, int nb_samples, int bytes_per_sample)
int frame_size
Number of samples per channel in an audio frame.
static av_cold int libopus_encode_init(AVCodecContext *avctx)
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
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)
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
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.
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
static const uint8_t opus_vorbis_channel_map[8][8]
common internal api header.
channel
Use these values when setting the channel map with ebur128_set_channel().
static const AVCodecDefault libopus_defaults[]
static av_always_inline void bytestream_put_buffer(uint8_t **b, const uint8_t *src, unsigned int size)
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
int cutoff
Audio cutoff bandwidth (0 means "automatic")
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[]
uint8_t * av_packet_new_side_data(AVPacket *pkt, enum AVPacketSideDataType type, int size)
Allocate new information of a packet.
const uint8_t ff_vorbis_channel_layout_offsets[8][8]
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...