23 #include <opus_multistream.h> 43 #ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST 48 #define OPUS_HEAD_SIZE 19 53 int ret, channel_map = 0, gain_db = 0,
nb_streams, nb_coupled;
54 uint8_t mapping_arr[8] = { 0, 1 }, *mapping;
59 "Invalid number of channels %d, defaulting to stereo\n", avc->
channels);
81 if (avc->
channels > 2 || channel_map) {
83 "No channel mapping for %d channels.\n", avc->
channels);
88 mapping = mapping_arr;
96 for (ch = 0; ch < avc->
channels; ch++)
97 mapping_arr[ch] = mapping[vorbis_offset[ch]];
98 mapping = mapping_arr;
111 ret = opus_multistream_decoder_ctl(opus->
dec, OPUS_SET_GAIN(gain_db));
117 double gain_lin =
ff_exp10(gain_db / (20.0 * 256));
119 opus->
gain.
d = gain_lin;
121 opus->
gain.
i =
FFMIN(gain_lin * 65536, INT_MAX);
125 #ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST 126 ret = opus_multistream_decoder_ctl(opus->
dec,
127 OPUS_SET_PHASE_INVERSION_DISABLED(!opus->apply_phase_inv));
130 "Unable to set phase inversion: %s\n",
145 opus_multistream_decoder_destroy(opus->
dec);
151 #define MAX_FRAME_SIZE (960 * 6) 165 nb_samples = opus_multistream_decode(opus->
dec, pkt->
data, pkt->
size,
166 (opus_int16 *)frame->
data[0],
169 nb_samples = opus_multistream_decode_float(opus->
dec, pkt->
data, pkt->
size,
170 (
float *)frame->
data[0],
173 if (nb_samples < 0) {
175 opus_strerror(nb_samples));
179 #ifndef OPUS_SET_GAIN 183 float *pcm = (
float *)frame->
data[0];
184 for (; i > 0; i--, pcm++)
185 *pcm = av_clipf(*pcm * opus->
gain.
d, -1, 1);
187 int16_t *pcm = (int16_t *)frame->
data[0];
188 for (; i > 0; i--, pcm++)
189 *pcm = av_clip_int16(((int64_t)opus->
gain.
i * *pcm) >> 16);
204 opus_multistream_decoder_ctl(opus->
dec, OPUS_RESET_STATE);
211 #define OFFSET(x) offsetof(struct libopus_context, x) 212 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM 214 #ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST 215 {
"apply_phase_inv",
"Apply intensity stereo phase inversion",
OFFSET(apply_phase_inv),
AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1,
FLAGS },
244 .wrapper_name =
"libopus",
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
static int libopus_decode(AVCodecContext *avc, void *data, int *got_frame_ptr, AVPacket *pkt)
This structure describes decoded (raw) audio or video data.
#define AV_LOG_WARNING
Something somehow does not look correct.
#define LIBAVUTIL_VERSION_INT
const char * av_default_item_name(void *ptr)
Return the context name.
AVCodec ff_libopus_decoder
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
int ff_opus_error_to_averror(int err)
enum AVSampleFormat sample_fmt
audio sample format
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
static av_always_inline double ff_exp10(double x)
Compute 10^x for floating point values.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
enum AVSampleFormat request_sample_fmt
desired sample format
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
const char * name
Name of the codec implementation.
uint64_t channel_layout
Audio channel layout.
common internal API header
static av_cold int libopus_decode_init(AVCodecContext *avc)
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
int sample_rate
samples per second
main external API structure.
const uint64_t ff_vorbis_channel_layouts[9]
static void libopus_flush(AVCodecContext *avc)
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Describe the class of an AVClass context structure.
static const AVClass libopusdec_class
int skip_samples
Number of audio samples to skip at the start of the next decoded frame.
static av_const int sign_extend(int val, unsigned bits)
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
internal math functions header
common internal api header.
static const AVOption libopusdec_options[]
int channels
number of audio channels
struct AVCodecInternal * internal
Private context used for internal data.
static av_cold int libopus_decode_close(AVCodecContext *avc)
union libopus_context::@88 gain
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
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.