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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-11-26 19:01:44 +02:00
FFmpeg/libavcodec/libspeexenc.c
Michael Niedermayer 305e4b35ea Merge remote-tracking branch 'qatar/master'
* qatar/master: (34 commits)
  mlp_parser: fix the channel mask value used for the top surround channel
  vorbisenc: check all allocations for failure
  roqaudioenc: return AVERROR codes instead of -1
  roqaudioenc: set correct bit rate
  roqaudioenc: use AVCodecContext.frame_size correctly.
  roqaudioenc: remove unneeded sample_fmt check
  ra144enc: use int16_t* for input samples rather than void*
  ra144enc: set AVCodecContext.coded_frame
  ra144enc: remove unneeded sample_fmt check
  nellymoserenc: set AVCodecContext.coded_frame
  nellymoserenc: improve error checking in encode_init()
  nellymoserenc: return AVERROR codes instead of -1
  libvorbis: improve error checking in oggvorbis_encode_init()
  mpegaudioenc: return AVERROR codes instead of -1
  libfaac: improve error checking and handling in Faac_encode_init()
  avutil: add AVERROR_UNKNOWN
  check for coded_frame allocation failure in several audio encoders
  audio encoders: do not set coded_frame->key_frame.
  g722enc: check for trellis data allocation error
  libspeexenc: export encoder delay through AVCodecContext.delay
  ...

Conflicts:
	doc/APIchanges
	libavcodec/avcodec.h
	libavcodec/fraps.c
	libavcodec/kgv1dec.c
	libavcodec/libfaac.c
	libavcodec/libgsm.c
	libavcodec/libvorbis.c
	libavcodec/mlp_parser.c
	libavcodec/roqaudioenc.c
	libavcodec/vorbisenc.c
	libavutil/avutil.h
	libavutil/error.c
	libavutil/error.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-02-26 05:11:21 +01:00

323 lines
12 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 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.
*/
#include <speex/speex.h>
#include <speex/speex_header.h>
#include <speex/speex_stereo.h>
#include "libavutil/opt.h"
#include "avcodec.h"
#include "internal.h"
typedef struct {
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 pkt_frame_count; ///< frame count for the current packet
int64_t next_pts; ///< next pts, in sample_rate time base
int pkt_sample_count; ///< sample count in the current packet
} 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: %d bps\n", avctx->bit_rate);
} else {
av_log(avctx, AV_LOG_DEBUG, "rate control: CBR\n");
av_log(avctx, AV_LOG_DEBUG, " bitrate: %d 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);
}
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_nb_mode; break;
case 16000: mode = &speex_wb_mode; break;
case 32000: mode = &speex_uwb_mode; 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 & CODEC_FLAG_QSCALE) {
/* VBR */
s->header.vbr = 1;
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);
}
/* 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->delay);
/* 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 and coded_frame */
avctx->extradata = av_malloc(header_size + FF_INPUT_BUFFER_PADDING_SIZE);
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->extradata || !avctx->coded_frame) {
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, uint8_t *frame, int buf_size,
void *data)
{
LibSpeexEncContext *s = avctx->priv_data;
int16_t *samples = data;
if (data) {
/* 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++;
s->pkt_sample_count += avctx->frame_size;
} 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;
avctx->coded_frame->pts = ff_samples_to_time_base(avctx, s->next_pts -
avctx->delay);
s->next_pts += s->pkt_sample_count;
s->pkt_sample_count = 0;
if (buf_size > speex_bits_nbytes(&s->bits)) {
int ret = speex_bits_write(&s->bits, frame, buf_size);
speex_bits_reset(&s->bits);
return ret;
} else {
av_log(avctx, AV_LOG_ERROR, "output buffer too small");
return AVERROR(EINVAL);
}
}
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);
av_freep(&avctx->coded_frame);
av_freep(&avctx->extradata);
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, { 0 }, 0, 1, AE },
{ "cbr_quality", "Set quality value (0 to 10) for CBR", OFFSET(cbr_quality), AV_OPT_TYPE_INT, { 8 }, 0, 10, AE },
{ "frames_per_packet", "Number of frames to encode in each packet", OFFSET(frames_per_packet), AV_OPT_TYPE_INT, { 1 }, 1, 8, AE },
{ NULL },
};
static const AVClass 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 },
};
AVCodec ff_libspeex_encoder = {
.name = "libspeex",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_SPEEX,
.priv_data_size = sizeof(LibSpeexEncContext),
.init = encode_init,
.encode = encode_frame,
.close = encode_close,
.capabilities = CODEC_CAP_DELAY,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE },
.long_name = NULL_IF_CONFIG_SMALL("libspeex Speex"),
.priv_class = &class,
.defaults = defaults,
};