diff options
Diffstat (limited to 'media/ffvpx/libavcodec')
| -rw-r--r-- | media/ffvpx/libavcodec/audio_frame_queue.c | 113 | ||||
| -rw-r--r-- | media/ffvpx/libavcodec/audio_frame_queue.h | 83 | ||||
| -rw-r--r-- | media/ffvpx/libavcodec/codec_list.c | 6 | ||||
| -rw-r--r-- | media/ffvpx/libavcodec/libopusenc.c | 610 | ||||
| -rw-r--r-- | media/ffvpx/libavcodec/libvorbisenc.c | 393 | ||||
| -rw-r--r-- | media/ffvpx/libavcodec/moz.build | 3 |
6 files changed, 1208 insertions, 0 deletions
diff --git a/media/ffvpx/libavcodec/audio_frame_queue.c b/media/ffvpx/libavcodec/audio_frame_queue.c new file mode 100644 index 0000000000..08b4b368c7 --- /dev/null +++ b/media/ffvpx/libavcodec/audio_frame_queue.c @@ -0,0 +1,113 @@ +/* + * Audio Frame Queue + * Copyright (c) 2012 Justin Ruggles + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "libavutil/attributes.h" +#include "libavutil/common.h" +#include "audio_frame_queue.h" +#include "encode.h" +#include "libavutil/avassert.h" + +av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq) +{ + afq->avctx = avctx; + afq->remaining_delay = avctx->initial_padding; + afq->remaining_samples = avctx->initial_padding; + afq->frame_count = 0; +} + +void ff_af_queue_close(AudioFrameQueue *afq) +{ + if(afq->frame_count) + av_log(afq->avctx, AV_LOG_WARNING, "%d frames left in the queue on closing\n", afq->frame_count); + av_freep(&afq->frames); + memset(afq, 0, sizeof(*afq)); +} + +int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f) +{ + AudioFrame *new = av_fast_realloc(afq->frames, &afq->frame_alloc, sizeof(*afq->frames)*(afq->frame_count+1)); + if(!new) + return AVERROR(ENOMEM); + afq->frames = new; + new += afq->frame_count; + + /* get frame parameters */ + new->duration = f->nb_samples; + new->duration += afq->remaining_delay; + if (f->pts != AV_NOPTS_VALUE) { + new->pts = av_rescale_q(f->pts, + afq->avctx->time_base, + (AVRational){ 1, afq->avctx->sample_rate }); + new->pts -= afq->remaining_delay; + if(afq->frame_count && new[-1].pts >= new->pts) + av_log(afq->avctx, AV_LOG_WARNING, "Queue input is backward in time\n"); + } else { + new->pts = AV_NOPTS_VALUE; + } + afq->remaining_delay = 0; + + /* add frame sample count */ + afq->remaining_samples += f->nb_samples; + + afq->frame_count++; + + return 0; +} + +void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, + int64_t *duration) +{ + int64_t out_pts = AV_NOPTS_VALUE; + int removed_samples = 0; + int i; + + if (afq->frame_count || afq->frame_alloc) { + if (afq->frames->pts != AV_NOPTS_VALUE) + out_pts = afq->frames->pts; + } + if(!afq->frame_count) + av_log(afq->avctx, AV_LOG_WARNING, "Trying to remove %d samples, but the queue is empty\n", nb_samples); + if (pts) + *pts = ff_samples_to_time_base(afq->avctx, out_pts); + + for(i=0; nb_samples && i<afq->frame_count; i++){ + int n= FFMIN(afq->frames[i].duration, nb_samples); + afq->frames[i].duration -= n; + nb_samples -= n; + removed_samples += n; + if(afq->frames[i].pts != AV_NOPTS_VALUE) + afq->frames[i].pts += n; + } + afq->remaining_samples -= removed_samples; + i -= i && afq->frames[i-1].duration; + memmove(afq->frames, afq->frames + i, sizeof(*afq->frames) * (afq->frame_count - i)); + afq->frame_count -= i; + + if(nb_samples){ + av_assert0(!afq->frame_count); + av_assert0(afq->remaining_samples == afq->remaining_delay); + if(afq->frames && afq->frames[0].pts != AV_NOPTS_VALUE) + afq->frames[0].pts += nb_samples; + av_log(afq->avctx, AV_LOG_DEBUG, "Trying to remove %d more samples than there are in the queue\n", nb_samples); + } + if (duration) + *duration = ff_samples_to_time_base(afq->avctx, removed_samples); +} diff --git a/media/ffvpx/libavcodec/audio_frame_queue.h b/media/ffvpx/libavcodec/audio_frame_queue.h new file mode 100644 index 0000000000..d8076eae54 --- /dev/null +++ b/media/ffvpx/libavcodec/audio_frame_queue.h @@ -0,0 +1,83 @@ +/* + * Audio Frame Queue + * Copyright (c) 2012 Justin Ruggles + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVCODEC_AUDIO_FRAME_QUEUE_H +#define AVCODEC_AUDIO_FRAME_QUEUE_H + +#include "avcodec.h" + +typedef struct AudioFrame { + int64_t pts; + int duration; +} AudioFrame; + +typedef struct AudioFrameQueue { + AVCodecContext *avctx; + int remaining_delay; + int remaining_samples; + AudioFrame *frames; + unsigned frame_count; + unsigned frame_alloc; +} AudioFrameQueue; + +/** + * Initialize AudioFrameQueue. + * + * @param avctx context to use for time_base and av_log + * @param afq queue context + */ +void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq); + +/** + * Close AudioFrameQueue. + * + * Frees memory if needed. + * + * @param afq queue context + */ +void ff_af_queue_close(AudioFrameQueue *afq); + +/** + * Add a frame to the queue. + * + * @param afq queue context + * @param f frame to add to the queue + */ +int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f); + +/** + * Remove frame(s) from the queue. + * + * Retrieves the pts of the next available frame, or a generated pts based on + * the last frame duration if there are no frames left in the queue. The number + * of requested samples should be the full number of samples represented by the + * packet that will be output by the encoder. If fewer samples are available + * in the queue, a smaller value will be used for the output duration. + * + * @param afq queue context + * @param nb_samples number of samples to remove from the queue + * @param[out] pts output packet pts + * @param[out] duration output packet duration + */ +void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, + int64_t *duration); + +#endif /* AVCODEC_AUDIO_FRAME_QUEUE_H */ diff --git a/media/ffvpx/libavcodec/codec_list.c b/media/ffvpx/libavcodec/codec_list.c index 04259e3cd7..7c6b0ceacd 100644 --- a/media/ffvpx/libavcodec/codec_list.c +++ b/media/ffvpx/libavcodec/codec_list.c @@ -20,6 +20,9 @@ static const FFCodec * const codec_list[] = { #if CONFIG_LIBVORBIS_DECODER &ff_libvorbis_decoder, #endif +#if CONFIG_LIBVORBIS_ENCODER + &ff_libvorbis_encoder, +#endif #if CONFIG_PCM_ALAW_DECODER &ff_pcm_alaw_decoder, #endif @@ -44,6 +47,9 @@ static const FFCodec * const codec_list[] = { #if CONFIG_LIBOPUS_DECODER &ff_libopus_decoder, #endif +#if CONFIG_LIBOPUS_ENCODER + &ff_libopus_encoder, +#endif #if CONFIG_LIBVPX_VP8_DECODER &ff_libvpx_vp8_decoder, #endif diff --git a/media/ffvpx/libavcodec/libopusenc.c b/media/ffvpx/libavcodec/libopusenc.c new file mode 100644 index 0000000000..68667e3350 --- /dev/null +++ b/media/ffvpx/libavcodec/libopusenc.c @@ -0,0 +1,610 @@ +/* + * Opus encoder using libopus + * Copyright (c) 2012 Nathan Caldwell + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <opus.h> +#include <opus_multistream.h> + +#include "libavutil/channel_layout.h" +#include "libavutil/opt.h" +#include "avcodec.h" +#include "bytestream.h" +#include "codec_internal.h" +#include "encode.h" +#include "libopus.h" +#include "audio_frame_queue.h" +#include "vorbis_data.h" + +typedef struct LibopusEncOpts { + int vbr; + int application; + int packet_loss; + int fec; + int complexity; + float frame_duration; + int packet_size; + int max_bandwidth; + int mapping_family; + int dtx; +#ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST + int apply_phase_inv; +#endif +} LibopusEncOpts; + +typedef struct LibopusEncContext { + AVClass *class; + OpusMSEncoder *enc; + int stream_count; + uint8_t *samples; + LibopusEncOpts opts; + AudioFrameQueue afq; + const uint8_t *encoder_channel_map; +} LibopusEncContext; + +static const uint8_t opus_coupled_streams[8] = { + 0, 1, 1, 2, 2, 2, 2, 3 +}; + +/* Opus internal to Vorbis channel order mapping written in the header */ +static const uint8_t opus_vorbis_channel_map[8][8] = { + { 0 }, + { 0, 1 }, + { 0, 2, 1 }, + { 0, 1, 2, 3 }, + { 0, 4, 1, 2, 3 }, + { 0, 4, 1, 2, 3, 5 }, + { 0, 4, 1, 2, 3, 5, 6 }, + { 0, 6, 1, 2, 3, 4, 5, 7 }, +}; + +/* libavcodec to libopus channel order mapping, passed to libopus */ +static const uint8_t libavcodec_libopus_channel_map[8][8] = { + { 0 }, + { 0, 1 }, + { 0, 1, 2 }, + { 0, 1, 2, 3 }, + { 0, 1, 3, 4, 2 }, + { 0, 1, 4, 5, 2, 3 }, + { 0, 1, 5, 6, 2, 4, 3 }, + { 0, 1, 6, 7, 4, 5, 2, 3 }, +}; + +static void libopus_write_header(AVCodecContext *avctx, int stream_count, + int coupled_stream_count, + int mapping_family, + const uint8_t *channel_mapping) +{ + uint8_t *p = avctx->extradata; + int channels = avctx->ch_layout.nb_channels; + + bytestream_put_buffer(&p, "OpusHead", 8); + bytestream_put_byte(&p, 1); /* Version */ + bytestream_put_byte(&p, channels); + bytestream_put_le16(&p, avctx->initial_padding * 48000 / avctx->sample_rate); /* Lookahead samples at 48kHz */ + bytestream_put_le32(&p, avctx->sample_rate); /* Original sample rate */ + bytestream_put_le16(&p, 0); /* Gain of 0dB is recommended. */ + + /* Channel mapping */ + bytestream_put_byte(&p, mapping_family); + if (mapping_family != 0) { + bytestream_put_byte(&p, stream_count); + bytestream_put_byte(&p, coupled_stream_count); + bytestream_put_buffer(&p, channel_mapping, channels); + } +} + +static int libopus_configure_encoder(AVCodecContext *avctx, OpusMSEncoder *enc, + LibopusEncOpts *opts) +{ + int ret; + + if (avctx->global_quality) { + av_log(avctx, AV_LOG_ERROR, + "Quality-based encoding not supported, " + "please specify a bitrate and VBR setting.\n"); + return AVERROR(EINVAL); + } + + ret = opus_multistream_encoder_ctl(enc, OPUS_SET_BITRATE(avctx->bit_rate)); + if (ret != OPUS_OK) { + av_log(avctx, AV_LOG_ERROR, + "Failed to set bitrate: %s\n", opus_strerror(ret)); + return ret; + } + + ret = opus_multistream_encoder_ctl(enc, + OPUS_SET_COMPLEXITY(opts->complexity)); + if (ret != OPUS_OK) + av_log(avctx, AV_LOG_WARNING, + "Unable to set complexity: %s\n", opus_strerror(ret)); + + ret = opus_multistream_encoder_ctl(enc, OPUS_SET_VBR(!!opts->vbr)); + if (ret != OPUS_OK) + av_log(avctx, AV_LOG_WARNING, + "Unable to set VBR: %s\n", opus_strerror(ret)); + + ret = opus_multistream_encoder_ctl(enc, + OPUS_SET_VBR_CONSTRAINT(opts->vbr == 2)); + if (ret != OPUS_OK) + av_log(avctx, AV_LOG_WARNING, + "Unable to set constrained VBR: %s\n", opus_strerror(ret)); + + ret = opus_multistream_encoder_ctl(enc, + OPUS_SET_PACKET_LOSS_PERC(opts->packet_loss)); + if (ret != OPUS_OK) + av_log(avctx, AV_LOG_WARNING, + "Unable to set expected packet loss percentage: %s\n", + opus_strerror(ret)); + + ret = opus_multistream_encoder_ctl(enc, + OPUS_SET_INBAND_FEC(opts->fec)); + if (ret != OPUS_OK) + av_log(avctx, AV_LOG_WARNING, + "Unable to set inband FEC: %s\n", + opus_strerror(ret)); + + ret = opus_multistream_encoder_ctl(enc, + OPUS_SET_DTX(opts->dtx)); + if (ret != OPUS_OK) + av_log(avctx, AV_LOG_WARNING, + "Unable to set DTX: %s\n", + opus_strerror(ret)); + + if (avctx->cutoff) { + ret = opus_multistream_encoder_ctl(enc, + OPUS_SET_MAX_BANDWIDTH(opts->max_bandwidth)); + if (ret != OPUS_OK) + av_log(avctx, AV_LOG_WARNING, + "Unable to set maximum bandwidth: %s\n", opus_strerror(ret)); + } + +#ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST + ret = opus_multistream_encoder_ctl(enc, + OPUS_SET_PHASE_INVERSION_DISABLED(!opts->apply_phase_inv)); + if (ret != OPUS_OK) + av_log(avctx, AV_LOG_WARNING, + "Unable to set phase inversion: %s\n", + opus_strerror(ret)); +#endif + return OPUS_OK; +} + +static int libopus_check_max_channels(AVCodecContext *avctx, + int max_channels) { + if (avctx->ch_layout.nb_channels > max_channels) { + av_log(avctx, AV_LOG_ERROR, "Opus mapping family undefined for %d channels.\n", + avctx->ch_layout.nb_channels); + return AVERROR(EINVAL); + } + + return 0; +} + +static int libopus_check_vorbis_layout(AVCodecContext *avctx, int mapping_family) { + av_assert2(avctx->ch_layout.nb_channels < FF_ARRAY_ELEMS(ff_vorbis_ch_layouts)); + + if (avctx->ch_layout.order == AV_CHANNEL_ORDER_UNSPEC) { + av_log(avctx, AV_LOG_WARNING, + "No channel layout specified. Opus encoder will use Vorbis " + "channel layout for %d channels.\n", avctx->ch_layout.nb_channels); + } else if (av_channel_layout_compare(&avctx->ch_layout, &ff_vorbis_ch_layouts[avctx->ch_layout.nb_channels - 1])) { + char name[32]; + + av_channel_layout_describe(&avctx->ch_layout, name, sizeof(name)); + av_log(avctx, AV_LOG_ERROR, + "Invalid channel layout %s for specified mapping family %d.\n", + name, mapping_family); + + return AVERROR(EINVAL); + } + + return 0; +} + +static int libopus_validate_layout_and_get_channel_map( + AVCodecContext *avctx, + int mapping_family, + const uint8_t ** channel_map_result) +{ + const uint8_t * channel_map = NULL; + int ret; + + switch (mapping_family) { + case -1: + ret = libopus_check_max_channels(avctx, 8); + if (ret == 0) { + ret = libopus_check_vorbis_layout(avctx, mapping_family); + /* Channels do not need to be reordered. */ + } + + break; + case 0: + ret = libopus_check_max_channels(avctx, 2); + if (ret == 0) { + ret = libopus_check_vorbis_layout(avctx, mapping_family); + } + break; + case 1: + /* Opus expects channels to be in Vorbis order. */ + ret = libopus_check_max_channels(avctx, 8); + if (ret == 0) { + ret = libopus_check_vorbis_layout(avctx, mapping_family); + channel_map = ff_vorbis_channel_layout_offsets[avctx->ch_layout.nb_channels - 1]; + } + break; + case 255: + ret = libopus_check_max_channels(avctx, 254); + break; + default: + av_log(avctx, AV_LOG_WARNING, + "Unknown channel mapping family %d. Output channel layout may be invalid.\n", + mapping_family); + ret = 0; + } + + *channel_map_result = channel_map; + return ret; +} + +static av_cold int libopus_encode_init(AVCodecContext *avctx) +{ + LibopusEncContext *opus = avctx->priv_data; + OpusMSEncoder *enc; + uint8_t libopus_channel_mapping[255]; + int ret = OPUS_OK; + int channels = avctx->ch_layout.nb_channels; + int av_ret; + int coupled_stream_count, header_size, frame_size; + int mapping_family; + + frame_size = opus->opts.frame_duration * 48000 / 1000; + switch (frame_size) { + case 120: + case 240: + if (opus->opts.application != OPUS_APPLICATION_RESTRICTED_LOWDELAY) + av_log(avctx, AV_LOG_WARNING, + "LPC mode cannot be used with a frame duration of less " + "than 10ms. Enabling restricted low-delay mode.\n" + "Use a longer frame duration if this is not what you want.\n"); + /* Frame sizes less than 10 ms can only use MDCT mode, so switching to + * RESTRICTED_LOWDELAY avoids an unnecessary extra 2.5ms lookahead. */ + opus->opts.application = OPUS_APPLICATION_RESTRICTED_LOWDELAY; + case 480: + case 960: + case 1920: + case 2880: +#ifdef OPUS_FRAMESIZE_120_MS + case 3840: + case 4800: + case 5760: +#endif + opus->opts.packet_size = + avctx->frame_size = frame_size * avctx->sample_rate / 48000; + break; + default: + av_log(avctx, AV_LOG_ERROR, "Invalid frame duration: %g.\n" + "Frame duration must be exactly one of: 2.5, 5, 10, 20, 40" +#ifdef OPUS_FRAMESIZE_120_MS + ", 60, 80, 100 or 120.\n", +#else + " or 60.\n", +#endif + opus->opts.frame_duration); + return AVERROR(EINVAL); + } + + if (avctx->compression_level < 0 || avctx->compression_level > 10) { + av_log(avctx, AV_LOG_WARNING, + "Compression level must be in the range 0 to 10. " + "Defaulting to 10.\n"); + opus->opts.complexity = 10; + } else { + opus->opts.complexity = avctx->compression_level; + } + + if (avctx->cutoff) { + switch (avctx->cutoff) { + case 4000: + opus->opts.max_bandwidth = OPUS_BANDWIDTH_NARROWBAND; + break; + case 6000: + opus->opts.max_bandwidth = OPUS_BANDWIDTH_MEDIUMBAND; + break; + case 8000: + opus->opts.max_bandwidth = OPUS_BANDWIDTH_WIDEBAND; + break; + case 12000: + opus->opts.max_bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND; + break; + case 20000: + opus->opts.max_bandwidth = OPUS_BANDWIDTH_FULLBAND; + break; + default: + av_log(avctx, AV_LOG_WARNING, + "Invalid frequency cutoff: %d. Using default maximum bandwidth.\n" + "Cutoff frequency must be exactly one of: 4000, 6000, 8000, 12000 or 20000.\n", + avctx->cutoff); + avctx->cutoff = 0; + } + } + + /* Channels may need to be reordered to match opus mapping. */ + av_ret = libopus_validate_layout_and_get_channel_map(avctx, opus->opts.mapping_family, + &opus->encoder_channel_map); + if (av_ret) { + return av_ret; + } + + if (opus->opts.mapping_family == -1) { + /* By default, use mapping family 1 for the header but use the older + * libopus multistream API to avoid surround masking. */ + + /* Set the mapping family so that the value is correct in the header */ + mapping_family = channels > 2 ? 1 : 0; + coupled_stream_count = opus_coupled_streams[channels - 1]; + opus->stream_count = channels - coupled_stream_count; + memcpy(libopus_channel_mapping, + opus_vorbis_channel_map[channels - 1], + channels * sizeof(*libopus_channel_mapping)); + + enc = opus_multistream_encoder_create( + avctx->sample_rate, channels, opus->stream_count, + coupled_stream_count, + libavcodec_libopus_channel_map[channels - 1], + opus->opts.application, &ret); + } else { + /* Use the newer multistream API. The encoder will set the channel + * mapping and coupled stream counts to its internal defaults and will + * use surround masking analysis to save bits. */ + mapping_family = opus->opts.mapping_family; + enc = opus_multistream_surround_encoder_create( + avctx->sample_rate, channels, mapping_family, + &opus->stream_count, &coupled_stream_count, libopus_channel_mapping, + opus->opts.application, &ret); + } + + if (ret != OPUS_OK) { + av_log(avctx, AV_LOG_ERROR, + "Failed to create encoder: %s\n", opus_strerror(ret)); + return ff_opus_error_to_averror(ret); + } + + if (!avctx->bit_rate) { + /* Sane default copied from opusenc */ + avctx->bit_rate = 64000 * opus->stream_count + + 32000 * coupled_stream_count; + av_log(avctx, AV_LOG_WARNING, + "No bit rate set. Defaulting to %"PRId64" bps.\n", avctx->bit_rate); + } + + if (avctx->bit_rate < 500 || avctx->bit_rate > 256000 * channels) { + av_log(avctx, AV_LOG_ERROR, "The bit rate %"PRId64" bps is unsupported. " + "Please choose a value between 500 and %d.\n", avctx->bit_rate, + 256000 * channels); + ret = AVERROR(EINVAL); + goto fail; + } + + ret = libopus_configure_encoder(avctx, enc, &opus->opts); + if (ret != OPUS_OK) { + ret = ff_opus_error_to_averror(ret); + goto fail; + } + + /* Header includes channel mapping table if and only if mapping family is NOT 0 */ + header_size = 19 + (mapping_family == 0 ? 0 : 2 + channels); + avctx->extradata = av_malloc(header_size + AV_INPUT_BUFFER_PADDING_SIZE); + if (!avctx->extradata) { + av_log(avctx, AV_LOG_ERROR, "Failed to allocate extradata.\n"); + ret = AVERROR(ENOMEM); + goto fail; + } + avctx->extradata_size = header_size; + + opus->samples = av_calloc(frame_size, channels * + av_get_bytes_per_sample(avctx->sample_fmt)); + if (!opus->samples) { + av_log(avctx, AV_LOG_ERROR, "Failed to allocate samples buffer.\n"); + ret = AVERROR(ENOMEM); + goto fail; + } + + ret = opus_multistream_encoder_ctl(enc, OPUS_GET_LOOKAHEAD(&avctx->initial_padding)); + if (ret != OPUS_OK) + av_log(avctx, AV_LOG_WARNING, + "Unable to get number of lookahead samples: %s\n", + opus_strerror(ret)); + + libopus_write_header(avctx, opus->stream_count, coupled_stream_count, + mapping_family, libopus_channel_mapping); + + ff_af_queue_init(avctx, &opus->afq); + + opus->enc = enc; + + return 0; + +fail: + opus_multistream_encoder_destroy(enc); + return ret; +} + +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 sample, channel; + for (sample = 0; sample < nb_samples; ++sample) { + for (channel = 0; channel < nb_channels; ++channel) { + const size_t src_pos = bytes_per_sample * (nb_channels * sample + channel); + const size_t dst_pos = bytes_per_sample * (nb_channels * sample + channel_map[channel]); + + memcpy(&dst[dst_pos], &src[src_pos], bytes_per_sample); + } + } +} + +static int libopus_encode(AVCodecContext *avctx, AVPacket *avpkt, + const AVFrame *frame, int *got_packet_ptr) +{ + LibopusEncContext *opus = avctx->priv_data; + const int bytes_per_sample = av_get_bytes_per_sample(avctx->sample_fmt); + const int channels = avctx->ch_layout.nb_channels; + const int sample_size = channels * bytes_per_sample; + const uint8_t *audio; + int ret; + int discard_padding; + + if (frame) { + ret = ff_af_queue_add(&opus->afq, frame); + if (ret < 0) + return ret; + if (opus->encoder_channel_map != NULL) { + audio = opus->samples; + libopus_copy_samples_with_channel_map( + opus->samples, frame->data[0], opus->encoder_channel_map, + channels, frame->nb_samples, bytes_per_sample); + } else if (frame->nb_samples < opus->opts.packet_size) { + audio = opus->samples; + memcpy(opus->samples, frame->data[0], frame->nb_samples * sample_size); + } else + audio = frame->data[0]; + } else { + if (!opus->afq.remaining_samples || (!opus->afq.frame_alloc && !opus->afq.frame_count)) + return 0; + audio = opus->samples; + memset(opus->samples, 0, opus->opts.packet_size * sample_size); + } + + /* Maximum packet size taken from opusenc in opus-tools. 120ms packets + * consist of 6 frames in one packet. The maximum frame size is 1275 + * bytes along with the largest possible packet header of 7 bytes. */ + if ((ret = ff_alloc_packet(avctx, avpkt, (1275 * 6 + 7) * opus->stream_count)) < 0) + return ret; + + if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) + ret = opus_multistream_encode_float(opus->enc, (const float *)audio, + opus->opts.packet_size, + avpkt->data, avpkt->size); + else + ret = opus_multistream_encode(opus->enc, (const opus_int16 *)audio, + opus->opts.packet_size, + avpkt->data, avpkt->size); + + if (ret < 0) { + av_log(avctx, AV_LOG_ERROR, + "Error encoding frame: %s\n", opus_strerror(ret)); + return ff_opus_error_to_averror(ret); + } + + av_shrink_packet(avpkt, ret); + + ff_af_queue_remove(&opus->afq, opus->opts.packet_size, + &avpkt->pts, &avpkt->duration); + + discard_padding = opus->opts.packet_size - avpkt->duration; + // Check if subtraction resulted in an overflow + if ((discard_padding < opus->opts.packet_size) != (avpkt->duration > 0)) + return AVERROR(EINVAL); + if (discard_padding > 0) { + uint8_t* side_data = av_packet_new_side_data(avpkt, + AV_PKT_DATA_SKIP_SAMPLES, + 10); + if (!side_data) + return AVERROR(ENOMEM); + AV_WL32(side_data + 4, discard_padding); + } + + *got_packet_ptr = 1; + + return 0; +} + +static av_cold int libopus_encode_close(AVCodecContext *avctx) +{ + LibopusEncContext *opus = avctx->priv_data; + + opus_multistream_encoder_destroy(opus->enc); + + ff_af_queue_close(&opus->afq); + + av_freep(&opus->samples); + + return 0; +} + +#define OFFSET(x) offsetof(LibopusEncContext, opts.x) +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM +static const AVOption libopus_options[] = { + { "application", "Intended application type", OFFSET(application), AV_OPT_TYPE_INT, { .i64 = OPUS_APPLICATION_AUDIO }, OPUS_APPLICATION_VOIP, OPUS_APPLICATION_RESTRICTED_LOWDELAY, FLAGS, "application" }, + { "voip", "Favor improved speech intelligibility", 0, AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_VOIP }, 0, 0, FLAGS, "application" }, + { "audio", "Favor faithfulness to the input", 0, AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_AUDIO }, 0, 0, FLAGS, "application" }, + { "lowdelay", "Restrict to only the lowest delay modes", 0, AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_RESTRICTED_LOWDELAY }, 0, 0, FLAGS, "application" }, + { "frame_duration", "Duration of a frame in milliseconds", OFFSET(frame_duration), AV_OPT_TYPE_FLOAT, { .dbl = 20.0 }, 2.5, 120.0, FLAGS }, + { "packet_loss", "Expected packet loss percentage", OFFSET(packet_loss), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 100, FLAGS }, + { "fec", "Enable inband FEC. Expected packet loss must be non-zero", OFFSET(fec), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, FLAGS }, + { "vbr", "Variable bit rate mode", OFFSET(vbr), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 2, FLAGS, "vbr" }, + { "off", "Use constant bit rate", 0, AV_OPT_TYPE_CONST, { .i64 = 0 }, 0, 0, FLAGS, "vbr" }, + { "on", "Use variable bit rate", 0, AV_OPT_TYPE_CONST, { .i64 = 1 }, 0, 0, FLAGS, "vbr" }, + { "constrained", "Use constrained VBR", 0, AV_OPT_TYPE_CONST, { .i64 = 2 }, 0, 0, FLAGS, "vbr" }, + { "mapping_family", "Channel Mapping Family", OFFSET(mapping_family), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 255, FLAGS, "mapping_family" }, + { "dtx", "Enable DTX", OFFSET(dtx), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, FLAGS }, +#ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST + { "apply_phase_inv", "Apply intensity stereo phase inversion", OFFSET(apply_phase_inv), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, FLAGS }, +#endif + { NULL }, +}; + +static const AVClass libopus_class = { + .class_name = "libopus", + .item_name = av_default_item_name, + .option = libopus_options, + .version = LIBAVUTIL_VERSION_INT, +}; + +static const FFCodecDefault libopus_defaults[] = { + { "b", "0" }, + { "compression_level", "10" }, + { NULL }, +}; + +static const int libopus_sample_rates[] = { + 48000, 24000, 16000, 12000, 8000, 0, +}; + +const FFCodec ff_libopus_encoder = { + .p.name = "libopus", + CODEC_LONG_NAME("libopus Opus"), + .p.type = AVMEDIA_TYPE_AUDIO, + .p.id = AV_CODEC_ID_OPUS, + .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY | + AV_CODEC_CAP_SMALL_LAST_FRAME, + .caps_internal = FF_CODEC_CAP_NOT_INIT_THREADSAFE, + .priv_data_size = sizeof(LibopusEncContext), + .init = libopus_encode_init, + FF_CODEC_ENCODE_CB(libopus_encode), + .close = libopus_encode_close, + .p.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, + AV_SAMPLE_FMT_FLT, + AV_SAMPLE_FMT_NONE }, + .p.supported_samplerates = libopus_sample_rates, + .p.priv_class = &libopus_class, + .defaults = libopus_defaults, + .p.wrapper_name = "libopus", +}; diff --git a/media/ffvpx/libavcodec/libvorbisenc.c b/media/ffvpx/libavcodec/libvorbisenc.c new file mode 100644 index 0000000000..6331cf0d79 --- /dev/null +++ b/media/ffvpx/libavcodec/libvorbisenc.c @@ -0,0 +1,393 @@ +/* + * Copyright (c) 2002 Mark Hills <mark@pogo.org.uk> + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <vorbis/vorbisenc.h> + +#include "libavutil/avassert.h" +#include "libavutil/channel_layout.h" +#include "libavutil/fifo.h" +#include "libavutil/opt.h" +#include "avcodec.h" +#include "audio_frame_queue.h" +#include "codec_internal.h" +#include "encode.h" +#include "version.h" +#include "vorbis_parser.h" + + +/* Number of samples the user should send in each call. + * This value is used because it is the LCD of all possible frame sizes, so + * an output packet will always start at the same point as one of the input + * packets. + */ +#define LIBVORBIS_FRAME_SIZE 64 + +#define BUFFER_SIZE (1024 * 64) + +typedef struct LibvorbisEncContext { + AVClass *av_class; /**< class for AVOptions */ + vorbis_info vi; /**< vorbis_info used during init */ + vorbis_dsp_state vd; /**< DSP state used for analysis */ + vorbis_block vb; /**< vorbis_block used for analysis */ + AVFifo *pkt_fifo; /**< output packet buffer */ + int eof; /**< end-of-file flag */ + int dsp_initialized; /**< vd has been initialized */ + vorbis_comment vc; /**< VorbisComment info */ + double iblock; /**< impulse block bias option */ + AVVorbisParseContext *vp; /**< parse context to get durations */ + AudioFrameQueue afq; /**< frame queue for timestamps */ +} LibvorbisEncContext; + +static const AVOption options[] = { + { "iblock", "Sets the impulse block bias", offsetof(LibvorbisEncContext, iblock), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, -15, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM }, + { NULL } +}; + +static const FFCodecDefault defaults[] = { + { "b", "0" }, + { NULL }, +}; + +static const AVClass vorbis_class = { + .class_name = "libvorbis", + .item_name = av_default_item_name, + .option = options, + .version = LIBAVUTIL_VERSION_INT, +}; + +static const uint8_t vorbis_encoding_channel_layout_offsets[8][8] = { + { 0 }, + { 0, 1 }, + { 0, 2, 1 }, + { 0, 1, 2, 3 }, + { 0, 2, 1, 3, 4 }, + { 0, 2, 1, 4, 5, 3 }, + { 0, 2, 1, 5, 6, 4, 3 }, + { 0, 2, 1, 6, 7, 4, 5, 3 }, +}; + +static int vorbis_error_to_averror(int ov_err) +{ + switch (ov_err) { + case OV_EFAULT: return AVERROR_BUG; + case OV_EINVAL: return AVERROR(EINVAL); + case OV_EIMPL: return AVERROR(EINVAL); + default: return AVERROR_UNKNOWN; + } +} + +static av_cold int libvorbis_setup(vorbis_info *vi, AVCodecContext *avctx) +{ + LibvorbisEncContext *s = avctx->priv_data; + int channels = avctx->ch_layout.nb_channels; + double cfreq; + int ret; + + if (avctx->flags & AV_CODEC_FLAG_QSCALE || !avctx->bit_rate) { + /* variable bitrate + * NOTE: we use the oggenc range of -1 to 10 for global_quality for + * user convenience, but libvorbis uses -0.1 to 1.0. + */ + float q = avctx->global_quality / (float)FF_QP2LAMBDA; + /* default to 3 if the user did not set quality or bitrate */ + if (!(avctx->flags & AV_CODEC_FLAG_QSCALE)) + q = 3.0; + if ((ret = vorbis_encode_setup_vbr(vi, channels, + avctx->sample_rate, + q / 10.0))) + goto error; + } else { + int minrate = avctx->rc_min_rate > 0 ? avctx->rc_min_rate : -1; + int maxrate = avctx->rc_max_rate > 0 ? avctx->rc_max_rate : -1; + + /* average bitrate */ + if ((ret = vorbis_encode_setup_managed(vi, channels, + avctx->sample_rate, maxrate, + avctx->bit_rate, minrate))) + goto error; + + /* variable bitrate by estimate, disable slow rate management */ + if (minrate == -1 && maxrate == -1) + if ((ret = vorbis_encode_ctl(vi, OV_ECTL_RATEMANAGE2_SET, NULL))) + goto error; /* should not happen */ + } + + /* cutoff frequency */ + if (avctx->cutoff > 0) { + cfreq = avctx->cutoff / 1000.0; + if ((ret = vorbis_encode_ctl(vi, OV_ECTL_LOWPASS_SET, &cfreq))) + goto error; /* should not happen */ + } + + /* impulse block bias */ + if (s->iblock) { + if ((ret = vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &s->iblock))) + goto error; + } + + if ((channels == 3 && + av_channel_layout_compare(&avctx->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_SURROUND)) || + (channels == 4 && + av_channel_layout_compare(&avctx->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_2_2) && + av_channel_layout_compare(&avctx->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_QUAD)) || + (channels == 5 && + av_channel_layout_compare(&avctx->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT0) && + av_channel_layout_compare(&avctx->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT0_BACK)) || + (channels == 6 && + av_channel_layout_compare(&avctx->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT1) && + av_channel_layout_compare(&avctx->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT1_BACK)) || + (channels == 7 && + av_channel_layout_compare(&avctx->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_6POINT1)) || + (channels == 8 && + av_channel_layout_compare(&avctx->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_7POINT1))) { + if (avctx->ch_layout.order != AV_CHANNEL_ORDER_UNSPEC) { + char name[32]; + av_channel_layout_describe(&avctx->ch_layout, name, sizeof(name)); + av_log(avctx, AV_LOG_ERROR, "%s not supported by Vorbis: " + "output stream will have incorrect " + "channel layout.\n", name); + } else { + av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The encoder " + "will use Vorbis channel layout for " + "%d channels.\n", channels); + } + } + + if ((ret = vorbis_encode_setup_init(vi))) + goto error; + + return 0; +error: + return vorbis_error_to_averror(ret); +} + +/* How many bytes are needed for a buffer of length 'l' */ +static int xiph_len(int l) +{ + return 1 + l / 255 + l; +} + +static av_cold int libvorbis_encode_close(AVCodecContext *avctx) +{ + LibvorbisEncContext *s = avctx->priv_data; + + /* notify vorbisenc this is EOF */ + if (s->dsp_initialized) + vorbis_analysis_wrote(&s->vd, 0); + + vorbis_block_clear(&s->vb); + vorbis_dsp_clear(&s->vd); + vorbis_info_clear(&s->vi); + + av_fifo_freep2(&s->pkt_fifo); + ff_af_queue_close(&s->afq); + + av_vorbis_parse_free(&s->vp); + + return 0; +} + +static av_cold int libvorbis_encode_init(AVCodecContext *avctx) +{ + LibvorbisEncContext *s = avctx->priv_data; + ogg_packet header, header_comm, header_code; + uint8_t *p; + unsigned int offset; + int ret; + + vorbis_info_init(&s->vi); + if ((ret = libvorbis_setup(&s->vi, avctx))) { + av_log(avctx, AV_LOG_ERROR, "encoder setup failed\n"); + goto error; + } + if ((ret = vorbis_analysis_init(&s->vd, &s->vi))) { + av_log(avctx, AV_LOG_ERROR, "analysis init failed\n"); + ret = vorbis_error_to_averror(ret); + goto error; + } + s->dsp_initialized = 1; + if ((ret = vorbis_block_init(&s->vd, &s->vb))) { + av_log(avctx, AV_LOG_ERROR, "dsp init failed\n"); + ret = vorbis_error_to_averror(ret); + goto error; + } + + vorbis_comment_init(&s->vc); + if (!(avctx->flags & AV_CODEC_FLAG_BITEXACT)) + vorbis_comment_add_tag(&s->vc, "encoder", LIBAVCODEC_IDENT); + + if ((ret = vorbis_analysis_headerout(&s->vd, &s->vc, &header, &header_comm, + &header_code))) { + ret = vorbis_error_to_averror(ret); + goto error; + } + + avctx->extradata_size = 1 + xiph_len(header.bytes) + + xiph_len(header_comm.bytes) + + header_code.bytes; + p = avctx->extradata = av_malloc(avctx->extradata_size + + AV_INPUT_BUFFER_PADDING_SIZE); + if (!p) { + ret = AVERROR(ENOMEM); + goto error; + } + p[0] = 2; + offset = 1; + offset += av_xiphlacing(&p[offset], header.bytes); + offset += av_xiphlacing(&p[offset], header_comm.bytes); + memcpy(&p[offset], header.packet, header.bytes); + offset += header.bytes; + memcpy(&p[offset], header_comm.packet, header_comm.bytes); + offset += header_comm.bytes; + memcpy(&p[offset], header_code.packet, header_code.bytes); + offset += header_code.bytes; + av_assert0(offset == avctx->extradata_size); + + s->vp = av_vorbis_parse_init(avctx->extradata, avctx->extradata_size); + if (!s->vp) { + av_log(avctx, AV_LOG_ERROR, "invalid extradata\n"); + return ret; + } + + vorbis_comment_clear(&s->vc); + + avctx->frame_size = LIBVORBIS_FRAME_SIZE; + ff_af_queue_init(avctx, &s->afq); + + s->pkt_fifo = av_fifo_alloc2(BUFFER_SIZE, 1, 0); + if (!s->pkt_fifo) { + ret = AVERROR(ENOMEM); + goto error; + } + + return 0; +error: + libvorbis_encode_close(avctx); + return ret; +} + +static int libvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, + const AVFrame *frame, int *got_packet_ptr) +{ + LibvorbisEncContext *s = avctx->priv_data; + ogg_packet op; + int ret, duration; + + /* send samples to libvorbis */ + if (frame) { + const int samples = frame->nb_samples; + float **buffer; + int c, channels = s->vi.channels; + + buffer = vorbis_analysis_buffer(&s->vd, samples); + for (c = 0; c < channels; c++) { + int co = (channels > 8) ? c : + vorbis_encoding_channel_layout_offsets[channels - 1][c]; + memcpy(buffer[c], frame->extended_data[co], + samples * sizeof(*buffer[c])); + } + if ((ret = vorbis_analysis_wrote(&s->vd, samples)) < 0) { + av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n"); + return vorbis_error_to_averror(ret); + } + if ((ret = ff_af_queue_add(&s->afq, frame)) < 0) + return ret; + } else { + if (!s->eof && s->afq.frame_alloc) + if ((ret = vorbis_analysis_wrote(&s->vd, 0)) < 0) { + av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n"); + return vorbis_error_to_averror(ret); + } + s->eof = 1; + } + + /* retrieve available packets from libvorbis */ + while ((ret = vorbis_analysis_blockout(&s->vd, &s->vb)) == 1) { + if ((ret = vorbis_analysis(&s->vb, NULL)) < 0) + break; + if ((ret = vorbis_bitrate_addblock(&s->vb)) < 0) + break; + + /* add any available packets to the output packet buffer */ + while ((ret = vorbis_bitrate_flushpacket(&s->vd, &op)) == 1) { + if (av_fifo_can_write(s->pkt_fifo) < sizeof(ogg_packet) + op.bytes) { + av_log(avctx, AV_LOG_ERROR, "packet buffer is too small\n"); + return AVERROR_BUG; + } + av_fifo_write(s->pkt_fifo, &op, sizeof(ogg_packet)); + av_fifo_write(s->pkt_fifo, op.packet, op.bytes); + } + if (ret < 0) { + av_log(avctx, AV_LOG_ERROR, "error getting available packets\n"); + break; + } + } + if (ret < 0) { + av_log(avctx, AV_LOG_ERROR, "error getting available packets\n"); + return vorbis_error_to_averror(ret); + } + + /* Read an available packet if possible */ + if (av_fifo_read(s->pkt_fifo, &op, sizeof(ogg_packet)) < 0) + return 0; + + if ((ret = ff_get_encode_buffer(avctx, avpkt, op.bytes, 0)) < 0) + return ret; + av_fifo_read(s->pkt_fifo, avpkt->data, op.bytes); + + avpkt->pts = ff_samples_to_time_base(avctx, op.granulepos); + + duration = av_vorbis_parse_frame(s->vp, avpkt->data, avpkt->size); + if (duration > 0) { + /* we do not know encoder delay until we get the first packet from + * libvorbis, so we have to update the AudioFrameQueue counts */ + if (!avctx->initial_padding && s->afq.frames) { + avctx->initial_padding = duration; + av_assert0(!s->afq.remaining_delay); + s->afq.frames->duration += duration; + if (s->afq.frames->pts != AV_NOPTS_VALUE) + s->afq.frames->pts -= duration; + s->afq.remaining_samples += duration; + } + ff_af_queue_remove(&s->afq, duration, &avpkt->pts, &avpkt->duration); + } + + *got_packet_ptr = 1; + return 0; +} + +const FFCodec ff_libvorbis_encoder = { + .p.name = "libvorbis", + CODEC_LONG_NAME("libvorbis"), + .p.type = AVMEDIA_TYPE_AUDIO, + .p.id = AV_CODEC_ID_VORBIS, + .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY | + AV_CODEC_CAP_SMALL_LAST_FRAME, + .caps_internal = FF_CODEC_CAP_NOT_INIT_THREADSAFE, + .priv_data_size = sizeof(LibvorbisEncContext), + .init = libvorbis_encode_init, + FF_CODEC_ENCODE_CB(libvorbis_encode_frame), + .close = libvorbis_encode_close, + .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP, + AV_SAMPLE_FMT_NONE }, + .p.priv_class = &vorbis_class, + .defaults = defaults, + .p.wrapper_name = "libvorbis", +}; diff --git a/media/ffvpx/libavcodec/moz.build b/media/ffvpx/libavcodec/moz.build index 0ba603d172..886fa7a2cb 100644 --- a/media/ffvpx/libavcodec/moz.build +++ b/media/ffvpx/libavcodec/moz.build @@ -20,6 +20,7 @@ LOCAL_INCLUDES += ['/modules/fdlibm/inexact-math-override'] SharedLibrary('mozavcodec') SOURCES += [ 'allcodecs.c', + 'audio_frame_queue.c', 'avcodec.c', 'avdct.c', 'avfft.c', @@ -47,7 +48,9 @@ SOURCES += [ 'jrevdct.c', 'libopus.c', 'libopusdec.c', + 'libopusenc.c', 'libvorbisdec.c', + 'libvorbisenc.c', 'log2_tab.c', 'mpegaudio.c', 'mpegaudiodata.c', |