openharmony/third_party_ffmpeg
1
0
Fork 0
mirror of https://gitee.com/openharmony/third_party_ffmpeg synced 2024-11-23 19:30:05 +00:00
Code Issues Actions Packages Projects Releases Wiki Activity
4978850ca2
third_party_ffmpeg/libavcodec/libfdk-aacenc.c
Anton Khirnov 2df0c32ea1 lavc: use a separate field for exporting audio encoder padding 
Currently, the amount of padding inserted at the beginning by some audio
encoders, is exported through AVCodecContext.delay. However
- the term 'delay' is heavily overloaded and can have multiple different
  meanings even in the case of audio encoding.
- this field has entirely different meanings, depending on whether the
  codec context is used for encoding or decoding (and has yet another
  different meaning for video), preventing generic handling of the codec
  context.

Therefore, add a new field -- AVCodecContext.initial_padding. It could
conceivably be used for decoding as well at a later point.
2014-10-13 19:09:01 +00:00

431 lines
16 KiB
C
Raw Blame History

/*
* AAC encoder wrapper
* Copyright (c) 2012 Martin Storsjo
*
* This file is part of Libav.
*
* Permission to use, copy, modify, and/or distribute this software for any
* purpose with or without fee is hereby granted, provided that the above
* copyright notice and this permission notice appear in all copies.
*
* THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
* WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
* ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
* WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
* ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
* OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
#include <fdk-aac/aacenc_lib.h>
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "audio_frame_queue.h"
#include "internal.h"
typedef struct AACContext {
const AVClass *class;
HANDLE_AACENCODER handle;
int afterburner;
int eld_sbr;
int signaling;
int latm;
int header_period;
int vbr;
AudioFrameQueue afq;
} AACContext;
static const AVOption aac_enc_options[] = {
{ "afterburner", "Afterburner (improved quality)", offsetof(AACContext, afterburner), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
{ "eld_sbr", "Enable SBR for ELD (for SBR in other configurations, use the -profile parameter)", offsetof(AACContext, eld_sbr), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
{ "signaling", "SBR/PS signaling style", offsetof(AACContext, signaling), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 2, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
{ "default", "Choose signaling implicitly (explicit hierarchical by default, implicit if global header is disabled)", 0, AV_OPT_TYPE_CONST, { .i64 = -1 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
{ "implicit", "Implicit backwards compatible signaling", 0, AV_OPT_TYPE_CONST, { .i64 = 0 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
{ "explicit_sbr", "Explicit SBR, implicit PS signaling", 0, AV_OPT_TYPE_CONST, { .i64 = 1 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
{ "explicit_hierarchical", "Explicit hierarchical signaling", 0, AV_OPT_TYPE_CONST, { .i64 = 2 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
{ "latm", "Output LATM/LOAS encapsulated data", offsetof(AACContext, latm), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
{ "header_period", "StreamMuxConfig and PCE repetition period (in frames)", offsetof(AACContext, header_period), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 0xffff, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
{ "vbr", "VBR mode (1-5)", offsetof(AACContext, vbr), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 5, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
{ NULL }
};
static const AVClass aac_enc_class = {
"libfdk_aac", av_default_item_name, aac_enc_options, LIBAVUTIL_VERSION_INT
};
static const char *aac_get_error(AACENC_ERROR err)
{
switch (err) {
case AACENC_OK:
return "No error";
case AACENC_INVALID_HANDLE:
return "Invalid handle";
case AACENC_MEMORY_ERROR:
return "Memory allocation error";
case AACENC_UNSUPPORTED_PARAMETER:
return "Unsupported parameter";
case AACENC_INVALID_CONFIG:
return "Invalid config";
case AACENC_INIT_ERROR:
return "Initialization error";
case AACENC_INIT_AAC_ERROR:
return "AAC library initialization error";
case AACENC_INIT_SBR_ERROR:
return "SBR library initialization error";
case AACENC_INIT_TP_ERROR:
return "Transport library initialization error";
case AACENC_INIT_META_ERROR:
return "Metadata library initialization error";
case AACENC_ENCODE_ERROR:
return "Encoding error";
case AACENC_ENCODE_EOF:
return "End of file";
default:
return "Unknown error";
}
}
static int aac_encode_close(AVCodecContext *avctx)
{
AACContext *s = avctx->priv_data;
if (s->handle)
aacEncClose(&s->handle);
av_freep(&avctx->extradata);
ff_af_queue_close(&s->afq);
return 0;
}
static av_cold int aac_encode_init(AVCodecContext *avctx)
{
AACContext *s = avctx->priv_data;
int ret = AVERROR(EINVAL);
AACENC_InfoStruct info = { 0 };
CHANNEL_MODE mode;
AACENC_ERROR err;
int aot = FF_PROFILE_AAC_LOW + 1;
int sce = 0, cpe = 0;
if ((err = aacEncOpen(&s->handle, 0, avctx->channels)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to open the encoder: %s\n",
aac_get_error(err));
goto error;
}
if (avctx->profile != FF_PROFILE_UNKNOWN)
aot = avctx->profile + 1;
if ((err = aacEncoder_SetParam(s->handle, AACENC_AOT, aot)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to set the AOT %d: %s\n",
aot, aac_get_error(err));
goto error;
}
if (aot == FF_PROFILE_AAC_ELD + 1 && s->eld_sbr) {
if ((err = aacEncoder_SetParam(s->handle, AACENC_SBR_MODE,
1)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to enable SBR for ELD: %s\n",
aac_get_error(err));
goto error;
}
}
if ((err = aacEncoder_SetParam(s->handle, AACENC_SAMPLERATE,
avctx->sample_rate)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to set the sample rate %d: %s\n",
avctx->sample_rate, aac_get_error(err));
goto error;
}
switch (avctx->channels) {
case 1: mode = MODE_1; sce = 1; cpe = 0; break;
case 2: mode = MODE_2; sce = 0; cpe = 1; break;
case 3: mode = MODE_1_2; sce = 1; cpe = 1; break;
case 4: mode = MODE_1_2_1; sce = 2; cpe = 1; break;
case 5: mode = MODE_1_2_2; sce = 1; cpe = 2; break;
case 6: mode = MODE_1_2_2_1; sce = 2; cpe = 2; break;
/* The version macro is introduced the same time as the 7.1 support, so this
should suffice. */
#ifdef AACENCODER_LIB_VL0
case 8:
sce = 2;
cpe = 3;
if (avctx->channel_layout == AV_CH_LAYOUT_7POINT1) {
mode = MODE_7_1_REAR_SURROUND;
} else {
// MODE_1_2_2_2_1 and MODE_7_1_FRONT_CENTER use the same channel layout
mode = MODE_7_1_FRONT_CENTER;
}
break;
#endif
default:
av_log(avctx, AV_LOG_ERROR,
"Unsupported number of channels %d\n", avctx->channels);
goto error;
}
if ((err = aacEncoder_SetParam(s->handle, AACENC_CHANNELMODE,
mode)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR,
"Unable to set channel mode %d: %s\n", mode, aac_get_error(err));
goto error;
}
if ((err = aacEncoder_SetParam(s->handle, AACENC_CHANNELORDER,
1)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR,
"Unable to set wav channel order %d: %s\n",
mode, aac_get_error(err));
goto error;
}
if (avctx->flags & CODEC_FLAG_QSCALE || s->vbr) {
int mode = s->vbr ? s->vbr : avctx->global_quality;
if (mode < 1 || mode > 5) {
av_log(avctx, AV_LOG_WARNING,
"VBR quality %d out of range, should be 1-5\n", mode);
mode = av_clip(mode, 1, 5);
}
av_log(avctx, AV_LOG_WARNING,
"Note, the VBR setting is unsupported and only works with "
"some parameter combinations\n");
if ((err = aacEncoder_SetParam(s->handle, AACENC_BITRATEMODE,
mode)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to set the VBR bitrate mode %d: %s\n",
mode, aac_get_error(err));
goto error;
}
} else {
if (avctx->bit_rate <= 0) {
if (avctx->profile == FF_PROFILE_AAC_HE_V2) {
sce = 1;
cpe = 0;
}
avctx->bit_rate = (96*sce + 128*cpe) * avctx->sample_rate / 44;
if (avctx->profile == FF_PROFILE_AAC_HE ||
avctx->profile == FF_PROFILE_AAC_HE_V2 ||
avctx->profile == FF_PROFILE_MPEG2_AAC_HE ||
s->eld_sbr)
avctx->bit_rate /= 2;
}
if ((err = aacEncoder_SetParam(s->handle, AACENC_BITRATE,
avctx->bit_rate)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to set the bitrate %d: %s\n",
avctx->bit_rate, aac_get_error(err));
goto error;
}
}
/* Choose bitstream format - if global header is requested, use
* raw access units, otherwise use ADTS. */
if ((err = aacEncoder_SetParam(s->handle, AACENC_TRANSMUX,
avctx->flags & CODEC_FLAG_GLOBAL_HEADER ? 0 : s->latm ? 10 : 2)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to set the transmux format: %s\n",
aac_get_error(err));
goto error;
}
if (s->latm && s->header_period) {
if ((err = aacEncoder_SetParam(s->handle, AACENC_HEADER_PERIOD,
s->header_period)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to set header period: %s\n",
aac_get_error(err));
goto error;
}
}
/* If no signaling mode is chosen, use explicit hierarchical signaling
* if using mp4 mode (raw access units, with global header) and
* implicit signaling if using ADTS. */
if (s->signaling < 0)
s->signaling = avctx->flags & CODEC_FLAG_GLOBAL_HEADER ? 2 : 0;
if ((err = aacEncoder_SetParam(s->handle, AACENC_SIGNALING_MODE,
s->signaling)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to set signaling mode %d: %s\n",
s->signaling, aac_get_error(err));
goto error;
}
if ((err = aacEncoder_SetParam(s->handle, AACENC_AFTERBURNER,
s->afterburner)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to set afterburner to %d: %s\n",
s->afterburner, aac_get_error(err));
goto error;
}
if (avctx->cutoff > 0) {
if (avctx->cutoff < (avctx->sample_rate + 255) >> 8 || avctx->cutoff > 20000) {
av_log(avctx, AV_LOG_ERROR, "cutoff valid range is %d-20000\n",
(avctx->sample_rate + 255) >> 8);
goto error;
}
if ((err = aacEncoder_SetParam(s->handle, AACENC_BANDWIDTH,
avctx->cutoff)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to set the encoder bandwidth to %d: %s\n",
avctx->cutoff, aac_get_error(err));
goto error;
}
}
if ((err = aacEncEncode(s->handle, NULL, NULL, NULL, NULL)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to initialize the encoder: %s\n",
aac_get_error(err));
return AVERROR(EINVAL);
}
if ((err = aacEncInfo(s->handle, &info)) != AACENC_OK) {
av_log(avctx, AV_LOG_ERROR, "Unable to get encoder info: %s\n",
aac_get_error(err));
goto error;
}
avctx->frame_size = info.frameLength;
avctx->initial_padding = info.encoderDelay;
ff_af_queue_init(avctx, &s->afq);
if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) {
avctx->extradata_size = info.confSize;
avctx->extradata = av_mallocz(avctx->extradata_size +
FF_INPUT_BUFFER_PADDING_SIZE);
if (!avctx->extradata) {
ret = AVERROR(ENOMEM);
goto error;
}
memcpy(avctx->extradata, info.confBuf, info.confSize);
}
return 0;
error:
aac_encode_close(avctx);
return ret;
}
static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
AACContext *s = avctx->priv_data;
AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 };
AACENC_InArgs in_args = { 0 };
AACENC_OutArgs out_args = { 0 };
int in_buffer_identifier = IN_AUDIO_DATA;
int in_buffer_size, in_buffer_element_size;
int out_buffer_identifier = OUT_BITSTREAM_DATA;
int out_buffer_size, out_buffer_element_size;
void *in_ptr, *out_ptr;
int ret;
AACENC_ERROR err;
/* handle end-of-stream small frame and flushing */
if (!frame) {
in_args.numInSamples = -1;
} else {
in_ptr = frame->data[0];
in_buffer_size = 2 * avctx->channels * frame->nb_samples;
in_buffer_element_size = 2;
in_args.numInSamples = avctx->channels * frame->nb_samples;
in_buf.numBufs = 1;
in_buf.bufs = &in_ptr;
in_buf.bufferIdentifiers = &in_buffer_identifier;
in_buf.bufSizes = &in_buffer_size;
in_buf.bufElSizes = &in_buffer_element_size;
/* add current frame to the queue */
if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
return ret;
}
/* The maximum packet size is 6144 bits aka 768 bytes per channel. */
if ((ret = ff_alloc_packet(avpkt, FFMAX(8192, 768 * avctx->channels)))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
out_ptr = avpkt->data;
out_buffer_size = avpkt->size;
out_buffer_element_size = 1;
out_buf.numBufs = 1;
out_buf.bufs = &out_ptr;
out_buf.bufferIdentifiers = &out_buffer_identifier;
out_buf.bufSizes = &out_buffer_size;
out_buf.bufElSizes = &out_buffer_element_size;
if ((err = aacEncEncode(s->handle, &in_buf, &out_buf, &in_args,
&out_args)) != AACENC_OK) {
if (!frame && err == AACENC_ENCODE_EOF)
return 0;
av_log(avctx, AV_LOG_ERROR, "Unable to encode frame: %s\n",
aac_get_error(err));
return AVERROR(EINVAL);
}
if (!out_args.numOutBytes)
return 0;
/* Get the next frame pts & duration */
ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
&avpkt->duration);
avpkt->size = out_args.numOutBytes;
*got_packet_ptr = 1;
return 0;
}
static const AVProfile profiles[] = {
{ FF_PROFILE_AAC_LOW, "LC" },
{ FF_PROFILE_AAC_HE, "HE-AAC" },
{ FF_PROFILE_AAC_HE_V2, "HE-AACv2" },
{ FF_PROFILE_AAC_LD, "LD" },
{ FF_PROFILE_AAC_ELD, "ELD" },
{ FF_PROFILE_UNKNOWN },
};
static const AVCodecDefault aac_encode_defaults[] = {
{ "b", "0" },
{ NULL }
};
static const uint64_t aac_channel_layout[] = {
AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_STEREO,
AV_CH_LAYOUT_SURROUND,
AV_CH_LAYOUT_4POINT0,
AV_CH_LAYOUT_5POINT0_BACK,
AV_CH_LAYOUT_5POINT1_BACK,
#ifdef AACENCODER_LIB_VL0
AV_CH_LAYOUT_7POINT1_WIDE_BACK,
AV_CH_LAYOUT_7POINT1,
#endif
0,
};
static const int aac_sample_rates[] = {
96000, 88200, 64000, 48000, 44100, 32000,
24000, 22050, 16000, 12000, 11025, 8000, 0
};
AVCodec ff_libfdk_aac_encoder = {
.name = "libfdk_aac",
.long_name = NULL_IF_CONFIG_SMALL("Fraunhofer FDK AAC"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_AAC,
.priv_data_size = sizeof(AACContext),
.init = aac_encode_init,
.encode2 = aac_encode_frame,
.close = aac_encode_close,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.priv_class = &aac_enc_class,
.defaults = aac_encode_defaults,
.profiles = profiles,
.supported_samplerates = aac_sample_rates,
.channel_layouts = aac_channel_layout,
};
Reference in New Issue View Git Blame Copy Permalink