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Diffstat (limited to 'libavcodec/libvorbis.c')
-rw-r--r--libavcodec/libvorbis.c308
1 files changed, 170 insertions, 138 deletions
diff --git a/libavcodec/libvorbis.c b/libavcodec/libvorbis.c
index 19dfa34fcc..f5ad49e68f 100644
--- a/libavcodec/libvorbis.c
+++ b/libavcodec/libvorbis.c
@@ -20,12 +20,13 @@
/**
* @file
- * Ogg Vorbis codec support via libvorbisenc.
+ * Vorbis encoding support via libvorbisenc.
* @author Mark Hills <mark@pogo.org.uk>
*/
#include <vorbis/vorbisenc.h>
+#include "libavutil/fifo.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "bytestream.h"
@@ -35,32 +36,41 @@
#undef NDEBUG
#include <assert.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 OGGVORBIS_FRAME_SIZE 64
#define BUFFER_SIZE (1024 * 64)
typedef struct OggVorbisContext {
- AVClass *av_class;
- vorbis_info vi;
- vorbis_dsp_state vd;
- vorbis_block vb;
- uint8_t buffer[BUFFER_SIZE];
- int buffer_index;
- int eof;
-
- /* decoder */
- vorbis_comment vc;
- ogg_packet op;
-
- double iblock;
+ 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 */
+ AVFifoBuffer *pkt_fifo; /**< output packet buffer */
+ int eof; /**< end-of-file flag */
+ int dsp_initialized; /**< vd has been initialized */
+ vorbis_comment vc; /**< VorbisComment info */
+ ogg_packet op; /**< ogg packet */
+ double iblock; /**< impulse block bias option */
} OggVorbisContext;
static const AVOption options[] = {
{ "iblock", "Sets the impulse block bias", offsetof(OggVorbisContext, iblock), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, -15, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
{ NULL }
};
+
+static const AVCodecDefault defaults[] = {
+ { "b", "0" },
+ { NULL },
+};
+
static const AVClass class = { "libvorbis", av_default_item_name, options, LIBAVUTIL_VERSION_INT };
+
static int vorbis_error_to_averror(int ov_err)
{
switch (ov_err) {
@@ -71,27 +81,34 @@ static int vorbis_error_to_averror(int ov_err)
}
}
-static av_cold int oggvorbis_init_encoder(vorbis_info *vi, AVCodecContext *avccontext)
+static av_cold int oggvorbis_init_encoder(vorbis_info *vi,
+ AVCodecContext *avctx)
{
- OggVorbisContext *context = avccontext->priv_data;
+ OggVorbisContext *s = avctx->priv_data;
double cfreq;
int ret;
- if (avccontext->flags & CODEC_FLAG_QSCALE) {
- /* variable bitrate */
- float q = avccontext->global_quality / (float)FF_QP2LAMBDA;
- if ((ret = vorbis_encode_setup_vbr(vi, avccontext->channels,
- avccontext->sample_rate,
+ if (avctx->flags & 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 & CODEC_FLAG_QSCALE))
+ q = 3.0;
+ if ((ret = vorbis_encode_setup_vbr(vi, avctx->channels,
+ avctx->sample_rate,
q / 10.0)))
goto error;
} else {
- int minrate = avccontext->rc_min_rate > 0 ? avccontext->rc_min_rate : -1;
- int maxrate = avccontext->rc_min_rate > 0 ? avccontext->rc_max_rate : -1;
+ int minrate = avctx->rc_min_rate > 0 ? avctx->rc_min_rate : -1;
+ int maxrate = avctx->rc_max_rate > 0 ? avctx->rc_max_rate : -1;
- /* constant bitrate */
- if ((ret = vorbis_encode_setup_managed(vi, avccontext->channels,
- avccontext->sample_rate, minrate,
- avccontext->bit_rate, maxrate)))
+ /* average bitrate */
+ if ((ret = vorbis_encode_setup_managed(vi, avctx->channels,
+ avctx->sample_rate, maxrate,
+ avctx->bit_rate, minrate)))
goto error;
/* variable bitrate by estimate, disable slow rate management */
@@ -101,43 +118,44 @@ static av_cold int oggvorbis_init_encoder(vorbis_info *vi, AVCodecContext *avcco
}
/* cutoff frequency */
- if (avccontext->cutoff > 0) {
- cfreq = avccontext->cutoff / 1000.0;
+ 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 */
}
- if (context->iblock) {
- if ((ret = vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &context->iblock)))
+ /* impulse block bias */
+ if (s->iblock) {
+ if ((ret = vorbis_encode_ctl(vi, OV_ECTL_IBLOCK_SET, &s->iblock)))
goto error;
}
- if (avccontext->channels == 3 &&
- avccontext->channel_layout != (AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER) ||
- avccontext->channels == 4 &&
- avccontext->channel_layout != AV_CH_LAYOUT_2_2 &&
- avccontext->channel_layout != AV_CH_LAYOUT_QUAD ||
- avccontext->channels == 5 &&
- avccontext->channel_layout != AV_CH_LAYOUT_5POINT0 &&
- avccontext->channel_layout != AV_CH_LAYOUT_5POINT0_BACK ||
- avccontext->channels == 6 &&
- avccontext->channel_layout != AV_CH_LAYOUT_5POINT1 &&
- avccontext->channel_layout != AV_CH_LAYOUT_5POINT1_BACK ||
- avccontext->channels == 7 &&
- avccontext->channel_layout != (AV_CH_LAYOUT_5POINT1|AV_CH_BACK_CENTER) ||
- avccontext->channels == 8 &&
- avccontext->channel_layout != AV_CH_LAYOUT_7POINT1) {
- if (avccontext->channel_layout) {
+ if (avctx->channels == 3 &&
+ avctx->channel_layout != (AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER) ||
+ avctx->channels == 4 &&
+ avctx->channel_layout != AV_CH_LAYOUT_2_2 &&
+ avctx->channel_layout != AV_CH_LAYOUT_QUAD ||
+ avctx->channels == 5 &&
+ avctx->channel_layout != AV_CH_LAYOUT_5POINT0 &&
+ avctx->channel_layout != AV_CH_LAYOUT_5POINT0_BACK ||
+ avctx->channels == 6 &&
+ avctx->channel_layout != AV_CH_LAYOUT_5POINT1 &&
+ avctx->channel_layout != AV_CH_LAYOUT_5POINT1_BACK ||
+ avctx->channels == 7 &&
+ avctx->channel_layout != (AV_CH_LAYOUT_5POINT1|AV_CH_BACK_CENTER) ||
+ avctx->channels == 8 &&
+ avctx->channel_layout != AV_CH_LAYOUT_7POINT1) {
+ if (avctx->channel_layout) {
char name[32];
- av_get_channel_layout_string(name, sizeof(name), avccontext->channels,
- avccontext->channel_layout);
- av_log(avccontext, AV_LOG_ERROR, "%s not supported by Vorbis: "
+ av_get_channel_layout_string(name, sizeof(name), avctx->channels,
+ avctx->channel_layout);
+ av_log(avctx, AV_LOG_ERROR, "%s not supported by Vorbis: "
"output stream will have incorrect "
"channel layout.\n", name);
} else {
- av_log(avccontext, AV_LOG_WARNING, "No channel layout specified. The encoder "
+ av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The encoder "
"will use Vorbis channel layout for "
- "%d channels.\n", avccontext->channels);
+ "%d channels.\n", avctx->channels);
}
}
@@ -155,59 +173,64 @@ static int xiph_len(int l)
return 1 + l / 255 + l;
}
-static av_cold int oggvorbis_encode_close(AVCodecContext *avccontext)
+static av_cold int oggvorbis_encode_close(AVCodecContext *avctx)
{
- OggVorbisContext *context = avccontext->priv_data;
-/* ogg_packet op ; */
+ OggVorbisContext *s = avctx->priv_data;
- vorbis_analysis_wrote(&context->vd, 0); /* notify vorbisenc this is EOF */
+ /* notify vorbisenc this is EOF */
+ if (s->dsp_initialized)
+ vorbis_analysis_wrote(&s->vd, 0);
- vorbis_block_clear(&context->vb);
- vorbis_dsp_clear(&context->vd);
- vorbis_info_clear(&context->vi);
+ vorbis_block_clear(&s->vb);
+ vorbis_dsp_clear(&s->vd);
+ vorbis_info_clear(&s->vi);
- av_freep(&avccontext->coded_frame);
- av_freep(&avccontext->extradata);
+ av_fifo_free(s->pkt_fifo);
+ av_freep(&avctx->coded_frame);
+ av_freep(&avctx->extradata);
return 0;
}
-static av_cold int oggvorbis_encode_init(AVCodecContext *avccontext)
+static av_cold int oggvorbis_encode_init(AVCodecContext *avctx)
{
- OggVorbisContext *context = avccontext->priv_data;
+ OggVorbisContext *s = avctx->priv_data;
ogg_packet header, header_comm, header_code;
uint8_t *p;
unsigned int offset;
int ret;
- vorbis_info_init(&context->vi);
- if ((ret = oggvorbis_init_encoder(&context->vi, avccontext))) {
- av_log(avccontext, AV_LOG_ERROR, "oggvorbis_encode_init: init_encoder failed\n");
+ vorbis_info_init(&s->vi);
+ if ((ret = oggvorbis_init_encoder(&s->vi, avctx))) {
+ av_log(avctx, AV_LOG_ERROR, "encoder setup failed\n");
goto error;
}
- if ((ret = vorbis_analysis_init(&context->vd, &context->vi))) {
+ 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;
}
- if ((ret = vorbis_block_init(&context->vd, &context->vb))) {
+ 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(&context->vc);
- vorbis_comment_add_tag(&context->vc, "encoder", LIBAVCODEC_IDENT);
+ vorbis_comment_init(&s->vc);
+ vorbis_comment_add_tag(&s->vc, "encoder", LIBAVCODEC_IDENT);
- if ((ret = vorbis_analysis_headerout(&context->vd, &context->vc, &header,
- &header_comm, &header_code))) {
+ if ((ret = vorbis_analysis_headerout(&s->vd, &s->vc, &header, &header_comm,
+ &header_code))) {
ret = vorbis_error_to_averror(ret);
goto error;
}
- avccontext->extradata_size =
- 1 + xiph_len(header.bytes) + xiph_len(header_comm.bytes) +
- header_code.bytes;
- p = avccontext->extradata =
- av_malloc(avccontext->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);
+ avctx->extradata_size = 1 + xiph_len(header.bytes) +
+ xiph_len(header_comm.bytes) +
+ header_code.bytes;
+ p = avctx->extradata = av_malloc(avctx->extradata_size +
+ FF_INPUT_BUFFER_PADDING_SIZE);
if (!p) {
ret = AVERROR(ENOMEM);
goto error;
@@ -222,100 +245,107 @@ static av_cold int oggvorbis_encode_init(AVCodecContext *avccontext)
offset += header_comm.bytes;
memcpy(&p[offset], header_code.packet, header_code.bytes);
offset += header_code.bytes;
- assert(offset == avccontext->extradata_size);
+ assert(offset == avctx->extradata_size);
-#if 0
- vorbis_block_clear(&context->vb);
- vorbis_dsp_clear(&context->vd);
- vorbis_info_clear(&context->vi);
-#endif
- vorbis_comment_clear(&context->vc);
+ vorbis_comment_clear(&s->vc);
- avccontext->frame_size = OGGVORBIS_FRAME_SIZE;
+ avctx->frame_size = OGGVORBIS_FRAME_SIZE;
+
+ s->pkt_fifo = av_fifo_alloc(BUFFER_SIZE);
+ if (!s->pkt_fifo) {
+ ret = AVERROR(ENOMEM);
+ goto error;
+ }
- avccontext->coded_frame = avcodec_alloc_frame();
- if (!avccontext->coded_frame) {
+ avctx->coded_frame = avcodec_alloc_frame();
+ if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
return 0;
error:
- oggvorbis_encode_close(avccontext);
+ oggvorbis_encode_close(avctx);
return ret;
}
-static int oggvorbis_encode_frame(AVCodecContext *avccontext,
- unsigned char *packets,
+static int oggvorbis_encode_frame(AVCodecContext *avctx, unsigned char *packets,
int buf_size, void *data)
{
- OggVorbisContext *context = avccontext->priv_data;
+ OggVorbisContext *s = avctx->priv_data;
ogg_packet op;
- signed short *audio = data;
- int l;
+ float *audio = data;
+ int pkt_size, ret;
+ /* send samples to libvorbis */
if (data) {
- const int samples = avccontext->frame_size;
+ const int samples = avctx->frame_size;
float **buffer;
- int c, channels = context->vi.channels;
+ int c, channels = s->vi.channels;
- buffer = vorbis_analysis_buffer(&context->vd, samples);
+ buffer = vorbis_analysis_buffer(&s->vd, samples);
for (c = 0; c < channels; c++) {
+ int i;
int co = (channels > 8) ? c :
ff_vorbis_encoding_channel_layout_offsets[channels - 1][c];
- for (l = 0; l < samples; l++)
- buffer[c][l] = audio[l * channels + co] / 32768.f;
+ for (i = 0; i < samples; i++)
+ buffer[c][i] = audio[i * channels + co];
+ }
+ 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);
}
- vorbis_analysis_wrote(&context->vd, samples);
} else {
- if (!context->eof)
- vorbis_analysis_wrote(&context->vd, 0);
- context->eof = 1;
+ if (!s->eof)
+ 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;
}
- while (vorbis_analysis_blockout(&context->vd, &context->vb) == 1) {
- vorbis_analysis(&context->vb, NULL);
- vorbis_bitrate_addblock(&context->vb);
-
- while (vorbis_bitrate_flushpacket(&context->vd, &op)) {
- /* i'd love to say the following line is a hack, but sadly it's
- * not, apparently the end of stream decision is in libogg. */
- if (op.bytes == 1 && op.e_o_s)
- continue;
- if (context->buffer_index + sizeof(ogg_packet) + op.bytes > BUFFER_SIZE) {
- av_log(avccontext, AV_LOG_ERROR, "libvorbis: buffer overflow.\n");
- return AVERROR(EINVAL);
+ /* 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_space(s->pkt_fifo) < sizeof(ogg_packet) + op.bytes) {
+ av_log(avctx, AV_LOG_ERROR, "packet buffer is too small\n");
+ return AVERROR_BUG;
}
- memcpy(context->buffer + context->buffer_index, &op, sizeof(ogg_packet));
- context->buffer_index += sizeof(ogg_packet);
- memcpy(context->buffer + context->buffer_index, op.packet, op.bytes);
- context->buffer_index += op.bytes;
-// av_log(avccontext, AV_LOG_DEBUG, "e%d / %d\n", context->buffer_index, op.bytes);
+ av_fifo_generic_write(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
+ av_fifo_generic_write(s->pkt_fifo, op.packet, op.bytes, NULL);
+ }
+ 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);
+ }
- l = 0;
- if (context->buffer_index) {
- ogg_packet *op2 = (ogg_packet *)context->buffer;
- op2->packet = context->buffer + sizeof(ogg_packet);
-
- l = op2->bytes;
- avccontext->coded_frame->pts = ff_samples_to_time_base(avccontext,
- op2->granulepos);
- //FIXME we should reorder the user supplied pts and not assume that they are spaced by 1/sample_rate
-
- if (l > buf_size) {
- av_log(avccontext, AV_LOG_ERROR, "libvorbis: buffer overflow.\n");
+ /* output then next packet from the output buffer, if available */
+ pkt_size = 0;
+ if (av_fifo_size(s->pkt_fifo) >= sizeof(ogg_packet)) {
+ av_fifo_generic_read(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
+ pkt_size = op.bytes;
+ // FIXME: we should use the user-supplied pts and duration
+ avctx->coded_frame->pts = ff_samples_to_time_base(avctx,
+ op.granulepos);
+ if (pkt_size > buf_size) {
+ av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
return AVERROR(EINVAL);
}
-
- memcpy(packets, op2->packet, l);
- context->buffer_index -= l + sizeof(ogg_packet);
- memmove(context->buffer, context->buffer + l + sizeof(ogg_packet), context->buffer_index);
-// av_log(avccontext, AV_LOG_DEBUG, "E%d\n", l);
+ av_fifo_generic_read(s->pkt_fifo, packets, pkt_size, NULL);
}
- return l;
+ return pkt_size;
}
AVCodec ff_libvorbis_encoder = {
@@ -327,7 +357,9 @@ AVCodec ff_libvorbis_encoder = {
.encode = oggvorbis_encode_frame,
.close = oggvorbis_encode_close,
.capabilities = CODEC_CAP_DELAY,
- .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE },
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
+ AV_SAMPLE_FMT_NONE },
.long_name = NULL_IF_CONFIG_SMALL("libvorbis Vorbis"),
.priv_class = &class,
+ .defaults = defaults,
};
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-01 14:06:52 +0000