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Bug 1749046 - Vendor libopusenc.c and libvorbisenc.c and support files in ffvpx. r=chunmin

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Differential Revision: https://phabricator.services.mozilla.com/D199524
This commit is contained in:
Paul Adenot 2024-03-27 14:16:31 +00:00
parent d58677ace3
commit d96ebfc92b
10 changed files with 1220 additions and 4 deletions
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16
config/external/gkcodecs/gkcodecs.symbols vendored
View File

@ -98,6 +98,8 @@ opus_packet_get_nb_frames
opus_packet_get_samples_per_frame
opus_packet_parse
opus_strerror
opus_multistream_encode_float
opus_multistream_surround_encoder_create
# libtheora symbols
th_comment_clear
th_comment_init
@ -127,6 +129,19 @@ vorbis_synthesis_init
vorbis_synthesis_pcmout
vorbis_synthesis_read
vorbis_synthesis_restart
vorbis_encode_ctl
vorbis_bitrate_addblock
vorbis_analysis_buffer
vorbis_analysis_blockout
vorbis_encode_setup_vbr
vorbis_analysis
vorbis_analysis_init
vorbis_analysis_headerout
vorbis_encode_setup_init
vorbis_encode_setup_managed
vorbis_comment_add_tag
vorbis_bitrate_flushpacket
vorbis_analysis_wrote
# libvpx symbols
#ifndef MOZ_SYSTEM_LIBVPX
vpx_codec_build_config
@ -156,3 +171,4 @@ vpx_img_alloc
vpx_img_free
vpx_img_wrap
#endif
#

4
media/ffvpx/config_components_audio_only.h
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@ -787,7 +787,7 @@
#define CONFIG_LIBMP3LAME_ENCODER 0
#define CONFIG_LIBOPENCORE_AMRNB_ENCODER 0
#define CONFIG_LIBOPENJPEG_ENCODER 0
#define CONFIG_LIBOPUS_ENCODER 0
#define CONFIG_LIBOPUS_ENCODER 1
#define CONFIG_LIBRAV1E_ENCODER 0
#define CONFIG_LIBSHINE_ENCODER 0
#define CONFIG_LIBSPEEX_ENCODER 0
@ -795,7 +795,7 @@
#define CONFIG_LIBTHEORA_ENCODER 0
#define CONFIG_LIBTWOLAME_ENCODER 0
#define CONFIG_LIBVO_AMRWBENC_ENCODER 0
#define CONFIG_LIBVORBIS_ENCODER 0
#define CONFIG_LIBVORBIS_ENCODER 1
#define CONFIG_LIBVPX_VP8_ENCODER 0
#define CONFIG_LIBVPX_VP9_ENCODER 0
#define CONFIG_LIBWEBP_ANIM_ENCODER 0

4
media/ffvpx/config_components_audio_video.h
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@ -810,7 +810,7 @@
#define CONFIG_LIBMP3LAME_ENCODER 0
#define CONFIG_LIBOPENCORE_AMRNB_ENCODER 0
#define CONFIG_LIBOPENJPEG_ENCODER 0
#define CONFIG_LIBOPUS_ENCODER 0
#define CONFIG_LIBOPUS_ENCODER 1
#define CONFIG_LIBRAV1E_ENCODER 0
#define CONFIG_LIBSHINE_ENCODER 0
#define CONFIG_LIBSPEEX_ENCODER 0
@ -818,7 +818,7 @@
#define CONFIG_LIBTHEORA_ENCODER 0
#define CONFIG_LIBTWOLAME_ENCODER 0
#define CONFIG_LIBVO_AMRWBENC_ENCODER 0
#define CONFIG_LIBVORBIS_ENCODER 0
#define CONFIG_LIBVORBIS_ENCODER 1
#define CONFIG_LIBVPX_VP8_ENCODER 1
#define CONFIG_LIBVPX_VP9_ENCODER 1
#define CONFIG_LIBWEBP_ANIM_ENCODER 0

113
media/ffvpx/libavcodec/audio_frame_queue.c Normal file
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@ -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);
}

83
media/ffvpx/libavcodec/audio_frame_queue.h Normal file
View File

@ -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 */

6
media/ffvpx/libavcodec/codec_list.c
View File

@ -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

601
media/ffvpx/libavcodec/libopusenc.c Normal file
View File

@ -0,0 +1,601 @@
/*
* 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;
#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));
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" },
#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",
};

393
media/ffvpx/libavcodec/libvorbisenc.c Normal file
View File

@ -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",
};

3
media/ffvpx/libavcodec/moz.build
View File

@ -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',

1
media/ffvpx/libavutil/avutil.symbols
View File

@ -92,6 +92,7 @@ av_fifo_alloc
av_fifo_alloc2
av_fifo_alloc_array
av_fifo_can_read
av_fifo_can_write
av_fifo_drain
av_fifo_drain2
av_fifo_free