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|
/* encavcodecaudio.c
Copyright (c) 2003-2015 HandBrake Team
This file is part of the HandBrake source code
Homepage: <http://handbrake.fr/>.
It may be used under the terms of the GNU General Public License v2.
For full terms see the file COPYING file or visit http://www.gnu.org/licenses/gpl-2.0.html
*/
#include "hb.h"
#include "hbffmpeg.h"
struct hb_work_private_s
{
hb_job_t * job;
AVCodecContext * context;
int out_discrete_channels;
int samples_per_frame;
unsigned long max_output_bytes;
unsigned long input_samples;
uint8_t * output_buf;
uint8_t * input_buf;
hb_list_t * list;
AVAudioResampleContext *avresample;
};
static int encavcodecaInit( hb_work_object_t *, hb_job_t * );
static int encavcodecaWork( hb_work_object_t *, hb_buffer_t **, hb_buffer_t ** );
static void encavcodecaClose( hb_work_object_t * );
hb_work_object_t hb_encavcodeca =
{
WORK_ENCAVCODEC_AUDIO,
"AVCodec Audio encoder (libavcodec)",
encavcodecaInit,
encavcodecaWork,
encavcodecaClose
};
static int encavcodecaInit(hb_work_object_t *w, hb_job_t *job)
{
AVCodec *codec;
AVCodecContext *context;
hb_audio_t *audio = w->audio;
hb_work_private_t *pv = calloc(1, sizeof(hb_work_private_t));
w->private_data = pv;
pv->job = job;
pv->list = hb_list_init();
// channel count, layout and matrix encoding
int matrix_encoding;
uint64_t channel_layout = hb_ff_mixdown_xlat(audio->config.out.mixdown,
&matrix_encoding);
pv->out_discrete_channels =
hb_mixdown_get_discrete_channel_count(audio->config.out.mixdown);
// default settings and options
AVDictionary *av_opts = NULL;
const char *codec_name = NULL;
enum AVCodecID codec_id = AV_CODEC_ID_NONE;
enum AVSampleFormat sample_fmt = AV_SAMPLE_FMT_FLTP;
int bits_per_raw_sample = 0;
int profile = FF_PROFILE_UNKNOWN;
// override with encoder-specific values
switch (audio->config.out.codec)
{
case HB_ACODEC_AC3:
codec_id = AV_CODEC_ID_AC3;
if (matrix_encoding != AV_MATRIX_ENCODING_NONE)
av_dict_set(&av_opts, "dsur_mode", "on", 0);
break;
case HB_ACODEC_FFEAC3:
codec_id = AV_CODEC_ID_EAC3;
if (matrix_encoding != AV_MATRIX_ENCODING_NONE)
av_dict_set(&av_opts, "dsur_mode", "on", 0);
break;
case HB_ACODEC_FDK_AAC:
case HB_ACODEC_FDK_HAAC:
codec_name = "libfdk_aac";
sample_fmt = AV_SAMPLE_FMT_S16;
bits_per_raw_sample = 16;
switch (audio->config.out.codec)
{
case HB_ACODEC_FDK_HAAC:
profile = FF_PROFILE_AAC_HE;
break;
default:
profile = FF_PROFILE_AAC_LOW;
break;
}
// Libav's libfdk-aac wrapper expects back channels for 5.1
// audio, and will error out unless we translate the layout
if (channel_layout == AV_CH_LAYOUT_5POINT1)
channel_layout = AV_CH_LAYOUT_5POINT1_BACK;
break;
case HB_ACODEC_FFAAC:
codec_name = "aac";
av_dict_set(&av_opts, "stereo_mode", "ms_off", 0);
break;
case HB_ACODEC_FFFLAC:
case HB_ACODEC_FFFLAC24:
codec_id = AV_CODEC_ID_FLAC;
switch (audio->config.out.codec)
{
case HB_ACODEC_FFFLAC24:
sample_fmt = AV_SAMPLE_FMT_S32;
bits_per_raw_sample = 24;
break;
default:
sample_fmt = AV_SAMPLE_FMT_S16;
bits_per_raw_sample = 16;
break;
}
break;
default:
hb_error("encavcodecaInit: unsupported codec (0x%x)",
audio->config.out.codec);
return 1;
}
if (codec_name != NULL)
{
codec = avcodec_find_encoder_by_name(codec_name);
if (codec == NULL)
{
hb_error("encavcodecaInit: avcodec_find_encoder_by_name(%s) failed",
codec_name);
return 1;
}
}
else
{
codec = avcodec_find_encoder(codec_id);
if (codec == NULL)
{
hb_error("encavcodecaInit: avcodec_find_encoder(%d) failed",
codec_id);
return 1;
}
}
// allocate the context and apply the settings
context = avcodec_alloc_context3(codec);
hb_ff_set_sample_fmt(context, codec, sample_fmt);
context->bits_per_raw_sample = bits_per_raw_sample;
context->profile = profile;
context->channel_layout = channel_layout;
context->channels = pv->out_discrete_channels;
context->sample_rate = audio->config.out.samplerate;
if (audio->config.out.bitrate > 0)
{
context->bit_rate = audio->config.out.bitrate * 1000;
}
else if (audio->config.out.quality >= 0)
{
context->global_quality = audio->config.out.quality * FF_QP2LAMBDA;
context->flags |= CODEC_FLAG_QSCALE;
if (audio->config.out.codec == HB_ACODEC_FDK_AAC ||
audio->config.out.codec == HB_ACODEC_FDK_HAAC)
{
char vbr[2];
snprintf(vbr, 2, "%.1g", audio->config.out.quality);
av_dict_set(&av_opts, "vbr", vbr, 0);
}
}
if (audio->config.out.compression_level >= 0)
{
context->compression_level = audio->config.out.compression_level;
}
// For some codecs, libav requires the following flag to be set
// so that it fills extradata with global header information.
// If this flag is not set, it inserts the data into each
// packet instead.
context->flags |= CODEC_FLAG_GLOBAL_HEADER;
if (hb_avcodec_open(context, codec, &av_opts, 0))
{
hb_error("encavcodecaInit: hb_avcodec_open() failed");
return 1;
}
// avcodec_open populates the opts dictionary with the
// things it didn't recognize.
AVDictionaryEntry *t = NULL;
while ((t = av_dict_get(av_opts, "", t, AV_DICT_IGNORE_SUFFIX)))
{
hb_log("encavcodecaInit: Unknown avcodec option %s", t->key);
}
av_dict_free(&av_opts);
pv->context = context;
audio->config.out.samples_per_frame =
pv->samples_per_frame = context->frame_size;
pv->input_samples = context->frame_size * context->channels;
pv->input_buf = malloc(pv->input_samples * sizeof(float));
// Some encoders in libav (e.g. fdk-aac) fail if the output buffer
// size is not some minumum value. 8K seems to be enough :(
pv->max_output_bytes = MAX(FF_MIN_BUFFER_SIZE,
(pv->input_samples *
av_get_bytes_per_sample(context->sample_fmt)));
// sample_fmt conversion
if (context->sample_fmt != AV_SAMPLE_FMT_FLT)
{
pv->output_buf = malloc(pv->max_output_bytes);
pv->avresample = avresample_alloc_context();
if (pv->avresample == NULL)
{
hb_error("encavcodecaInit: avresample_alloc_context() failed");
return 1;
}
av_opt_set_int(pv->avresample, "in_sample_fmt",
AV_SAMPLE_FMT_FLT, 0);
av_opt_set_int(pv->avresample, "out_sample_fmt",
context->sample_fmt, 0);
av_opt_set_int(pv->avresample, "in_channel_layout",
context->channel_layout, 0);
av_opt_set_int(pv->avresample, "out_channel_layout",
context->channel_layout, 0);
if (hb_audio_dither_is_supported(audio->config.out.codec))
{
// dithering needs the sample rate
av_opt_set_int(pv->avresample, "in_sample_rate",
context->sample_rate, 0);
av_opt_set_int(pv->avresample, "out_sample_rate",
context->sample_rate, 0);
av_opt_set_int(pv->avresample, "dither_method",
audio->config.out.dither_method, 0);
}
if (avresample_open(pv->avresample))
{
hb_error("encavcodecaInit: avresample_open() failed");
avresample_free(&pv->avresample);
return 1;
}
}
else
{
pv->avresample = NULL;
pv->output_buf = pv->input_buf;
}
if (context->extradata != NULL)
{
memcpy(w->config->extradata.bytes, context->extradata,
context->extradata_size);
w->config->extradata.length = context->extradata_size;
}
audio->config.out.delay = av_rescale_q(context->delay, context->time_base,
(AVRational){1, 90000});
return 0;
}
/***********************************************************************
* Close
***********************************************************************
*
**********************************************************************/
// Some encoders (e.g. flac) require a final NULL encode in order to
// finalize things.
static void Finalize(hb_work_object_t *w)
{
hb_work_private_t *pv = w->private_data;
// Finalize with NULL input needed by FLAC to generate md5sum
// in context extradata
// Prepare output packet
AVPacket pkt;
int got_packet;
hb_buffer_t *buf = hb_buffer_init(pv->max_output_bytes);
av_init_packet(&pkt);
pkt.data = buf->data;
pkt.size = buf->alloc;
avcodec_encode_audio2(pv->context, &pkt, NULL, &got_packet);
hb_buffer_close(&buf);
// Then we need to recopy the header since it was modified
if (pv->context->extradata != NULL)
{
memcpy(w->config->extradata.bytes, pv->context->extradata,
pv->context->extradata_size);
w->config->extradata.length = pv->context->extradata_size;
}
}
static void encavcodecaClose(hb_work_object_t * w)
{
hb_work_private_t * pv = w->private_data;
if (pv != NULL)
{
if (pv->context != NULL)
{
Finalize(w);
hb_deep_log(2, "encavcodeca: closing libavcodec");
if (pv->context->codec != NULL)
avcodec_flush_buffers(pv->context);
hb_avcodec_close(pv->context);
av_free( pv->context );
}
if (pv->output_buf != NULL)
{
free(pv->output_buf);
}
if (pv->input_buf != NULL && pv->input_buf != pv->output_buf)
{
free(pv->input_buf);
}
pv->output_buf = pv->input_buf = NULL;
if (pv->list != NULL)
{
hb_list_empty(&pv->list);
}
if (pv->avresample != NULL)
{
avresample_free(&pv->avresample);
}
free(pv);
w->private_data = NULL;
}
}
static hb_buffer_t* Encode(hb_work_object_t *w)
{
hb_work_private_t *pv = w->private_data;
hb_audio_t *audio = w->audio;
uint64_t pts, pos;
if (hb_list_bytes(pv->list) < pv->input_samples * sizeof(float))
{
return NULL;
}
hb_list_getbytes(pv->list, pv->input_buf, pv->input_samples * sizeof(float),
&pts, &pos);
// Prepare input frame
int out_linesize;
int out_size = av_samples_get_buffer_size(&out_linesize,
pv->context->channels,
pv->samples_per_frame,
pv->context->sample_fmt, 1);
AVFrame frame = { .nb_samples = pv->samples_per_frame, };
avcodec_fill_audio_frame(&frame,
pv->context->channels, pv->context->sample_fmt,
pv->output_buf, out_size, 1);
if (pv->avresample != NULL)
{
int in_linesize;
av_samples_get_buffer_size(&in_linesize, pv->context->channels,
frame.nb_samples, AV_SAMPLE_FMT_FLT, 1);
int out_samples = avresample_convert(pv->avresample,
frame.extended_data, out_linesize,
frame.nb_samples,
&pv->input_buf, in_linesize,
frame.nb_samples);
if (out_samples != pv->samples_per_frame)
{
// we're not doing sample rate conversion, so this shouldn't happen
hb_log("encavcodecaWork: avresample_convert() failed");
return NULL;
}
}
// Libav requires that timebase of audio frames be in sample_rate units
frame.pts = pts + (90000 * pos / (sizeof(float) *
pv->out_discrete_channels *
audio->config.out.samplerate));
frame.pts = av_rescale(frame.pts, pv->context->sample_rate, 90000);
// Prepare output packet
AVPacket pkt;
int got_packet;
hb_buffer_t *out = hb_buffer_init(pv->max_output_bytes);
av_init_packet(&pkt);
pkt.data = out->data;
pkt.size = out->alloc;
// Encode
int ret = avcodec_encode_audio2(pv->context, &pkt, &frame, &got_packet);
if (ret < 0)
{
hb_log("encavcodeca: avcodec_encode_audio failed");
hb_buffer_close(&out);
return NULL;
}
if (got_packet && pkt.size)
{
out->size = pkt.size;
// The output pts from libav is in context->time_base. Convert it back
// to our timebase.
out->s.start = av_rescale_q(pkt.pts, pv->context->time_base,
(AVRational){1, 90000});
out->s.duration = (double)90000 * pv->samples_per_frame /
audio->config.out.samplerate;
out->s.stop = out->s.start + out->s.duration;
out->s.type = AUDIO_BUF;
out->s.frametype = HB_FRAME_AUDIO;
}
else
{
hb_buffer_close(&out);
return Encode(w);
}
return out;
}
static hb_buffer_t * Flush( hb_work_object_t * w )
{
hb_buffer_t *first, *buf, *last;
first = last = buf = Encode( w );
while( buf )
{
last = buf;
buf->next = Encode( w );
buf = buf->next;
}
if( last )
{
last->next = hb_buffer_eof_init();
}
else
{
first = hb_buffer_eof_init();
}
return first;
}
/***********************************************************************
* Work
***********************************************************************
*
**********************************************************************/
static int encavcodecaWork( hb_work_object_t * w, hb_buffer_t ** buf_in,
hb_buffer_t ** buf_out )
{
hb_work_private_t * pv = w->private_data;
hb_buffer_t * in = *buf_in, * buf;
if (in->s.flags & HB_BUF_FLAG_EOF)
{
/* EOF on input - send it downstream & say we're done */
*buf_out = Flush( w );
return HB_WORK_DONE;
}
if ( pv->context == NULL || pv->context->codec == NULL )
{
// No encoder context. Nothing we can do.
return HB_WORK_OK;
}
hb_list_add( pv->list, in );
*buf_in = NULL;
*buf_out = buf = Encode( w );
while ( buf )
{
buf->next = Encode( w );
buf = buf->next;
}
return HB_WORK_OK;
}
|