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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-23 12:43:46 +02:00
FFmpeg/libavcodec/libspeexenc.c
Andreas Rheinhardt 56e9e0273a avcodec/encode: Always use intermediate buffer in ff_alloc_packet2()
Up until now, ff_alloc_packet2() has a min_size parameter:
It is supposed to be a lower bound on the final size of the packet
to allocate. If it is not too far from the upper bound (namely,
if it is at least half the upper bound), then ff_alloc_packet2()
already allocates the final, already refcounted packet; if it is
not, then the packet is not refcounted and its data only points to
a buffer owned by the AVCodecContext (in this case, the packet will
be made refcounted in encode_simple_internal() in libavcodec/encode.c).
The goal of this was to avoid data copies and intermediate buffers
if one has a precise lower bound.

Yet those encoders for which precise lower bounds exist have recently
been switched to ff_get_encode_buffer() (which automatically allocates
final buffers), leaving only two encoders to actually set the min_size
to something else than zero (namely aliaspixenc and hapenc). Both of
these encoders use a very low lower bound that is not helpful in any
nontrivial case.

This commit therefore removes the min_size parameter as well as the
codepath in ff_alloc_packet2() for the allocation of final buffers.
Furthermore, the function has been renamed to ff_alloc_packet() and
moved to encode.h alongside ff_get_encode_buffer().

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-06-08 12:52:50 +02:00

370 lines
14 KiB
C

/*
* Copyright (C) 2009 Justin Ruggles
* Copyright (c) 2009 Xuggle Incorporated
*
* 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
*/
/**
* @file
* libspeex Speex audio encoder
*
* Usage Guide
* This explains the values that need to be set prior to initialization in
* order to control various encoding parameters.
*
* Channels
* Speex only supports mono or stereo, so avctx->channels must be set to
* 1 or 2.
*
* Sample Rate / Encoding Mode
* Speex has 3 modes, each of which uses a specific sample rate.
* narrowband : 8 kHz
* wideband : 16 kHz
* ultra-wideband : 32 kHz
* avctx->sample_rate must be set to one of these 3 values. This will be
* used to set the encoding mode.
*
* Rate Control
* VBR mode is turned on by setting AV_CODEC_FLAG_QSCALE in avctx->flags.
* avctx->global_quality is used to set the encoding quality.
* For CBR mode, avctx->bit_rate can be used to set the constant bitrate.
* Alternatively, the 'cbr_quality' option can be set from 0 to 10 to set
* a constant bitrate based on quality.
* For ABR mode, set avctx->bit_rate and set the 'abr' option to 1.
* Approx. Bitrate Range:
* narrowband : 2400 - 25600 bps
* wideband : 4000 - 43200 bps
* ultra-wideband : 4400 - 45200 bps
*
* Complexity
* Encoding complexity is controlled by setting avctx->compression_level.
* The valid range is 0 to 10. A higher setting gives generally better
* quality at the expense of encoding speed. This does not affect the
* bit rate.
*
* Frames-per-Packet
* The encoder defaults to using 1 frame-per-packet. However, it is
* sometimes desirable to use multiple frames-per-packet to reduce the
* amount of container overhead. This can be done by setting the
* 'frames_per_packet' option to a value 1 to 8.
*
*
* Optional features
* Speex encoder supports several optional features, which can be useful
* for some conditions.
*
* Voice Activity Detection
* When enabled, voice activity detection detects whether the audio
* being encoded is speech or silence/background noise. VAD is always
* implicitly activated when encoding in VBR, so the option is only useful
* in non-VBR operation. In this case, Speex detects non-speech periods and
* encodes them with just enough bits to reproduce the background noise.
*
* Discontinuous Transmission (DTX)
* DTX is an addition to VAD/VBR operation, that makes it possible to stop transmitting
* completely when the background noise is stationary.
* In file-based operation only 5 bits are used for such frames.
*/
#include <speex/speex.h>
#include <speex/speex_header.h>
#include <speex/speex_stereo.h>
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "encode.h"
#include "internal.h"
#include "audio_frame_queue.h"
/* TODO: Think about converting abr, vad, dtx and such flags to a bit field */
typedef struct LibSpeexEncContext {
AVClass *class; ///< AVClass for private options
SpeexBits bits; ///< libspeex bitwriter context
SpeexHeader header; ///< libspeex header struct
void *enc_state; ///< libspeex encoder state
int frames_per_packet; ///< number of frames to encode in each packet
float vbr_quality; ///< VBR quality 0.0 to 10.0
int cbr_quality; ///< CBR quality 0 to 10
int abr; ///< flag to enable ABR
int vad; ///< flag to enable VAD
int dtx; ///< flag to enable DTX
int pkt_frame_count; ///< frame count for the current packet
AudioFrameQueue afq; ///< frame queue
} LibSpeexEncContext;
static av_cold void print_enc_params(AVCodecContext *avctx,
LibSpeexEncContext *s)
{
const char *mode_str = "unknown";
av_log(avctx, AV_LOG_DEBUG, "channels: %d\n", avctx->channels);
switch (s->header.mode) {
case SPEEX_MODEID_NB: mode_str = "narrowband"; break;
case SPEEX_MODEID_WB: mode_str = "wideband"; break;
case SPEEX_MODEID_UWB: mode_str = "ultra-wideband"; break;
}
av_log(avctx, AV_LOG_DEBUG, "mode: %s\n", mode_str);
if (s->header.vbr) {
av_log(avctx, AV_LOG_DEBUG, "rate control: VBR\n");
av_log(avctx, AV_LOG_DEBUG, " quality: %f\n", s->vbr_quality);
} else if (s->abr) {
av_log(avctx, AV_LOG_DEBUG, "rate control: ABR\n");
av_log(avctx, AV_LOG_DEBUG, " bitrate: %"PRId64" bps\n", avctx->bit_rate);
} else {
av_log(avctx, AV_LOG_DEBUG, "rate control: CBR\n");
av_log(avctx, AV_LOG_DEBUG, " bitrate: %"PRId64" bps\n", avctx->bit_rate);
}
av_log(avctx, AV_LOG_DEBUG, "complexity: %d\n",
avctx->compression_level);
av_log(avctx, AV_LOG_DEBUG, "frame size: %d samples\n",
avctx->frame_size);
av_log(avctx, AV_LOG_DEBUG, "frames per packet: %d\n",
s->frames_per_packet);
av_log(avctx, AV_LOG_DEBUG, "packet size: %d\n",
avctx->frame_size * s->frames_per_packet);
av_log(avctx, AV_LOG_DEBUG, "voice activity detection: %d\n", s->vad);
av_log(avctx, AV_LOG_DEBUG, "discontinuous transmission: %d\n", s->dtx);
}
static av_cold int encode_init(AVCodecContext *avctx)
{
LibSpeexEncContext *s = avctx->priv_data;
const SpeexMode *mode;
uint8_t *header_data;
int header_size;
int32_t complexity;
/* channels */
if (avctx->channels < 1 || avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR, "Invalid channels (%d). Only stereo and "
"mono are supported\n", avctx->channels);
return AVERROR(EINVAL);
}
/* sample rate and encoding mode */
switch (avctx->sample_rate) {
case 8000: mode = speex_lib_get_mode(SPEEX_MODEID_NB); break;
case 16000: mode = speex_lib_get_mode(SPEEX_MODEID_WB); break;
case 32000: mode = speex_lib_get_mode(SPEEX_MODEID_UWB); break;
default:
av_log(avctx, AV_LOG_ERROR, "Sample rate of %d Hz is not supported. "
"Resample to 8, 16, or 32 kHz.\n", avctx->sample_rate);
return AVERROR(EINVAL);
}
/* initialize libspeex */
s->enc_state = speex_encoder_init(mode);
if (!s->enc_state) {
av_log(avctx, AV_LOG_ERROR, "Error initializing libspeex\n");
return -1;
}
speex_init_header(&s->header, avctx->sample_rate, avctx->channels, mode);
/* rate control method and parameters */
if (avctx->flags & AV_CODEC_FLAG_QSCALE) {
/* VBR */
s->header.vbr = 1;
s->vad = 1; /* VAD is always implicitly activated for VBR */
speex_encoder_ctl(s->enc_state, SPEEX_SET_VBR, &s->header.vbr);
s->vbr_quality = av_clipf(avctx->global_quality / (float)FF_QP2LAMBDA,
0.0f, 10.0f);
speex_encoder_ctl(s->enc_state, SPEEX_SET_VBR_QUALITY, &s->vbr_quality);
} else {
s->header.bitrate = avctx->bit_rate;
if (avctx->bit_rate > 0) {
/* CBR or ABR by bitrate */
if (s->abr) {
speex_encoder_ctl(s->enc_state, SPEEX_SET_ABR,
&s->header.bitrate);
speex_encoder_ctl(s->enc_state, SPEEX_GET_ABR,
&s->header.bitrate);
} else {
speex_encoder_ctl(s->enc_state, SPEEX_SET_BITRATE,
&s->header.bitrate);
speex_encoder_ctl(s->enc_state, SPEEX_GET_BITRATE,
&s->header.bitrate);
}
} else {
/* CBR by quality */
speex_encoder_ctl(s->enc_state, SPEEX_SET_QUALITY,
&s->cbr_quality);
speex_encoder_ctl(s->enc_state, SPEEX_GET_BITRATE,
&s->header.bitrate);
}
/* stereo side information adds about 800 bps to the base bitrate */
/* TODO: this should be calculated exactly */
avctx->bit_rate = s->header.bitrate + (avctx->channels == 2 ? 800 : 0);
}
/* VAD is activated with VBR or can be turned on by itself */
if (s->vad)
speex_encoder_ctl(s->enc_state, SPEEX_SET_VAD, &s->vad);
/* Activating Discontinuous Transmission */
if (s->dtx) {
speex_encoder_ctl(s->enc_state, SPEEX_SET_DTX, &s->dtx);
if (!(s->abr || s->vad || s->header.vbr))
av_log(avctx, AV_LOG_WARNING, "DTX is not much of use without ABR, VAD or VBR\n");
}
/* set encoding complexity */
if (avctx->compression_level > FF_COMPRESSION_DEFAULT) {
complexity = av_clip(avctx->compression_level, 0, 10);
speex_encoder_ctl(s->enc_state, SPEEX_SET_COMPLEXITY, &complexity);
}
speex_encoder_ctl(s->enc_state, SPEEX_GET_COMPLEXITY, &complexity);
avctx->compression_level = complexity;
/* set packet size */
avctx->frame_size = s->header.frame_size;
s->header.frames_per_packet = s->frames_per_packet;
/* set encoding delay */
speex_encoder_ctl(s->enc_state, SPEEX_GET_LOOKAHEAD, &avctx->initial_padding);
ff_af_queue_init(avctx, &s->afq);
/* create header packet bytes from header struct */
/* note: libspeex allocates the memory for header_data, which is freed
below with speex_header_free() */
header_data = speex_header_to_packet(&s->header, &header_size);
/* allocate extradata */
avctx->extradata = av_malloc(header_size + AV_INPUT_BUFFER_PADDING_SIZE);
if (!avctx->extradata) {
speex_header_free(header_data);
speex_encoder_destroy(s->enc_state);
av_log(avctx, AV_LOG_ERROR, "memory allocation error\n");
return AVERROR(ENOMEM);
}
/* copy header packet to extradata */
memcpy(avctx->extradata, header_data, header_size);
avctx->extradata_size = header_size;
speex_header_free(header_data);
/* init libspeex bitwriter */
speex_bits_init(&s->bits);
print_enc_params(avctx, s);
return 0;
}
static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
LibSpeexEncContext *s = avctx->priv_data;
int16_t *samples = frame ? (int16_t *)frame->data[0] : NULL;
int ret;
if (samples) {
/* encode Speex frame */
if (avctx->channels == 2)
speex_encode_stereo_int(samples, s->header.frame_size, &s->bits);
speex_encode_int(s->enc_state, samples, &s->bits);
s->pkt_frame_count++;
if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
return ret;
} else {
/* handle end-of-stream */
if (!s->pkt_frame_count)
return 0;
/* add extra terminator codes for unused frames in last packet */
while (s->pkt_frame_count < s->frames_per_packet) {
speex_bits_pack(&s->bits, 15, 5);
s->pkt_frame_count++;
}
}
/* write output if all frames for the packet have been encoded */
if (s->pkt_frame_count == s->frames_per_packet) {
s->pkt_frame_count = 0;
if ((ret = ff_alloc_packet(avctx, avpkt, speex_bits_nbytes(&s->bits))) < 0)
return ret;
ret = speex_bits_write(&s->bits, avpkt->data, avpkt->size);
speex_bits_reset(&s->bits);
/* Get the next frame pts/duration */
ff_af_queue_remove(&s->afq, s->frames_per_packet * avctx->frame_size,
&avpkt->pts, &avpkt->duration);
avpkt->size = ret;
*got_packet_ptr = 1;
return 0;
}
return 0;
}
static av_cold int encode_close(AVCodecContext *avctx)
{
LibSpeexEncContext *s = avctx->priv_data;
speex_bits_destroy(&s->bits);
speex_encoder_destroy(s->enc_state);
ff_af_queue_close(&s->afq);
return 0;
}
#define OFFSET(x) offsetof(LibSpeexEncContext, x)
#define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
static const AVOption options[] = {
{ "abr", "Use average bit rate", OFFSET(abr), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AE },
{ "cbr_quality", "Set quality value (0 to 10) for CBR", OFFSET(cbr_quality), AV_OPT_TYPE_INT, { .i64 = 8 }, 0, 10, AE },
{ "frames_per_packet", "Number of frames to encode in each packet", OFFSET(frames_per_packet), AV_OPT_TYPE_INT, { .i64 = 1 }, 1, 8, AE },
{ "vad", "Voice Activity Detection", OFFSET(vad), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AE },
{ "dtx", "Discontinuous Transmission", OFFSET(dtx), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AE },
{ NULL },
};
static const AVClass speex_class = {
.class_name = "libspeex",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
static const AVCodecDefault defaults[] = {
{ "b", "0" },
{ "compression_level", "3" },
{ NULL },
};
const AVCodec ff_libspeex_encoder = {
.name = "libspeex",
.long_name = NULL_IF_CONFIG_SMALL("libspeex Speex"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_SPEEX,
.priv_data_size = sizeof(LibSpeexEncContext),
.init = encode_init,
.encode2 = encode_frame,
.close = encode_close,
.capabilities = AV_CODEC_CAP_DELAY,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.channel_layouts = (const uint64_t[]){ AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_STEREO,
0 },
.supported_samplerates = (const int[]){ 8000, 16000, 32000, 0 },
.priv_class = &speex_class,
.defaults = defaults,
.wrapper_name = "libspeex",
};