RepoMirrors/mpv
1
0
Fork 0
mirror of https://github.com/mpv-player/mpv synced 2024-12-30 02:52:10 +00:00
Code Issues Releases Wiki Activity

audio: add support for using non-interleaved audio from decoders directly

Browse Source 
Most libavcodec decoders output non-interleaved audio. Add direct
support for this, and remove the hack that repacked non-interleaved
audio back to packed audio.

Remove the minlen argument from the decoder callback. Instead of
forcing every decoder to have its own decode loop to fill the buffer
until minlen is reached, leave this to the caller. So if a decoder
doesn't return enough data, it's simply called again. (In future, I
even want to change it so that decoders don't read packets directly,
but instead the caller has to pass packets to the decoders. This fits
well with this change, because now the decoder callback typically
decodes at most one packet.)

ad_mpg123.c receives some heavy refactoring. The main problem is that
it wanted to handle format changes when there was no data in the decode
output buffer yet. This sounds reasonable, but actually it would write
data into a buffer prepared for old data, since the caller doesn't know
about the format change yet. (I.e. the best place for a format change
would be _after_ writing the last sample to the output buffer.) It's
possible that this code was not perfectly sane before this commit,
and perhaps lost one frame of data after a format change, but I didn't
confirm this. Trying to fix this, I ended up rewriting the decoding
and also the probing.
This commit is contained in:
wm4 2013-11-12 22:27:44 +01:00
parent 5388a0cd40
commit 22b3f522ca
10 changed files with 375 additions and 546 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

4
audio/decode/ad.h
View File

@ -24,6 +24,7 @@
#include "demux/demux.h"
#include "audio/format.h"
#include "audio/audio.h"
struct mp_decoder_list;
@ -35,8 +36,7 @@ struct ad_functions {
int (*init)(sh_audio_t *sh, const char *decoder);
void (*uninit)(sh_audio_t *sh);
int (*control)(sh_audio_t *sh, int cmd, void *arg);
int (*decode_audio)(sh_audio_t *sh, unsigned char *buffer, int minlen,
int maxlen);
int (*decode_audio)(sh_audio_t *sh, struct mp_audio *buffer, int maxlen);
};
enum ad_ctrl {

161
audio/decode/ad_lavc.c
View File

@ -36,25 +36,20 @@
#include "mpvcore/av_opts.h"
#include "ad.h"
#include "audio/reorder_ch.h"
#include "audio/fmt-conversion.h"
#include "compat/mpbswap.h"
#include "compat/libav.h"
struct priv {
AVCodecContext *avctx;
AVFrame *avframe;
uint8_t *output;
uint8_t *output_packed; // used by deplanarize to store packed audio samples
int output_left;
int unitsize;
struct mp_audio frame;
bool force_channel_map;
struct demux_packet *packet;
};
static void uninit(sh_audio_t *sh);
static int decode_audio(sh_audio_t *sh,unsigned char *buffer,int minlen,int maxlen);
static int decode_new_packet(struct sh_audio *sh);
#define OPT_BASE_STRUCT struct MPOpts
@ -150,22 +145,21 @@ static int preinit(sh_audio_t *sh)
return 1;
}
/* Prefer playing audio with the samplerate given in container data
* if available, but take number the number of channels and sample format
* from the codec, since if the codec isn't using the correct values for
* those everything breaks anyway.
*/
static int setup_format(sh_audio_t *sh_audio,
const AVCodecContext *lavc_context)
static int setup_format(sh_audio_t *sh_audio)
{
struct priv *priv = sh_audio->context;
int sample_format =
af_from_avformat(av_get_packed_sample_fmt(lavc_context->sample_fmt));
int samplerate = lavc_context->sample_rate;
// If not set, try container samplerate
AVCodecContext *lavc_context = priv->avctx;
int sample_format = af_from_avformat(lavc_context->sample_fmt);
if (!sample_format)
return -1;
int samplerate = lavc_context->sample_rate;
if (!samplerate && sh_audio->wf) {
// If not set, try container samplerate.
// (Maybe this can't happen, and it's an artifact from the past.)
samplerate = sh_audio->wf->nSamplesPerSec;
mp_tmsg(MSGT_DECAUDIO, MSGL_V, "ad_lavc: using container rate.\n");
mp_tmsg(MSGT_DECAUDIO, MSGL_WARN, "ad_lavc: using container rate.\n");
}
struct mp_chmap lavc_chmap;
@ -178,14 +172,9 @@ static int setup_format(sh_audio_t *sh_audio,
lavc_chmap = sh_audio->channels;
}
if (!mp_chmap_equals(&lavc_chmap, &sh_audio->channels) ||
samplerate != sh_audio->samplerate ||
sample_format != sh_audio->sample_format) {
sh_audio->channels = lavc_chmap;
sh_audio->samplerate = samplerate;
sh_audio->sample_format = sample_format;
return 1;
}
sh_audio->channels = lavc_chmap;
sh_audio->samplerate = samplerate;
sh_audio->sample_format = sample_format;
return 0;
}
@ -285,15 +274,12 @@ static int init(sh_audio_t *sh_audio, const char *decoder)
mp_msg(MSGT_DECAUDIO, MSGL_V, "INFO: libavcodec \"%s\" init OK!\n",
lavc_codec->name);
// Decode at least 1 byte: (to get header filled)
for (int tries = 0;;) {
int x = decode_audio(sh_audio, sh_audio->a_buffer, 1,
sh_audio->a_buffer_size);
if (x > 0) {
sh_audio->a_buffer_len = x;
// Decode at least 1 sample: (to get header filled)
for (int tries = 1; ; tries++) {
int x = decode_new_packet(sh_audio);
if (x >= 0 && ctx->frame.samples > 0)
break;
}
if (++tries >= 5) {
if (tries >= 5) {
mp_msg(MSGT_DECAUDIO, MSGL_ERR,
"ad_lavc: initial decode failed\n");
uninit(sh_audio);
@ -305,12 +291,6 @@ static int init(sh_audio_t *sh_audio, const char *decoder)
if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
sh_audio->i_bps = sh_audio->wf->nAvgBytesPerSec;
int af_sample_fmt =
af_from_avformat(av_get_packed_sample_fmt(lavc_context->sample_fmt));
if (af_sample_fmt == AF_FORMAT_UNKNOWN) {
uninit(sh_audio);
return 0;
}
return 1;
}
@ -338,7 +318,7 @@ static int control(sh_audio_t *sh, int cmd, void *arg)
switch (cmd) {
case ADCTRL_RESYNC_STREAM:
avcodec_flush_buffers(ctx->avctx);
ctx->output_left = 0;
ctx->frame.samples = 0;
talloc_free(ctx->packet);
ctx->packet = NULL;
return CONTROL_TRUE;
@ -346,29 +326,13 @@ static int control(sh_audio_t *sh, int cmd, void *arg)
return CONTROL_UNKNOWN;
}
static av_always_inline void deplanarize(struct sh_audio *sh)
{
struct priv *priv = sh->context;
uint8_t **planes = priv->avframe->extended_data;
size_t bps = av_get_bytes_per_sample(priv->avctx->sample_fmt);
size_t nb_samples = priv->avframe->nb_samples;
size_t channels = priv->avctx->channels;
size_t size = bps * nb_samples * channels;
if (talloc_get_size(priv->output_packed) != size)
priv->output_packed =
talloc_realloc_size(priv, priv->output_packed, size);
reorder_to_packed(priv->output_packed, planes, bps, channels, nb_samples);
priv->output = priv->output_packed;
}
static int decode_new_packet(struct sh_audio *sh)
{
struct priv *priv = sh->context;
AVCodecContext *avctx = priv->avctx;
priv->frame.samples = 0;
struct demux_packet *mpkt = priv->packet;
if (!mpkt)
mpkt = demux_read_packet(sh->gsh);
@ -384,7 +348,7 @@ static int decode_new_packet(struct sh_audio *sh)
if (mpkt->pts != MP_NOPTS_VALUE) {
sh->pts = mpkt->pts;
sh->pts_bytes = 0;
sh->pts_offset = 0;
}
int got_frame = 0;
int ret = avcodec_decode_audio4(avctx, priv->avframe, &got_frame, &pkt);
@ -409,58 +373,39 @@ static int decode_new_packet(struct sh_audio *sh)
}
if (!got_frame)
return 0;
uint64_t unitsize = (uint64_t)av_get_bytes_per_sample(avctx->sample_fmt) *
avctx->channels;
if (unitsize > 100000)
abort();
priv->unitsize = unitsize;
uint64_t output_left = unitsize * priv->avframe->nb_samples;
if (output_left > 500000000)
abort();
priv->output_left = output_left;
if (av_sample_fmt_is_planar(avctx->sample_fmt) && avctx->channels > 1) {
deplanarize(sh);
} else {
priv->output = priv->avframe->data[0];
}
mp_dbg(MSGT_DECAUDIO, MSGL_DBG2, "Decoded %d -> %d \n", in_len,
priv->output_left);
if (setup_format(sh) < 0)
return -1;
priv->frame.samples = priv->avframe->nb_samples;
mp_audio_set_format(&priv->frame, sh->sample_format);
mp_audio_set_channels(&priv->frame, &sh->channels);
priv->frame.rate = sh->samplerate;
for (int n = 0; n < priv->frame.num_planes; n++)
priv->frame.planes[n] = priv->avframe->data[n];
mp_dbg(MSGT_DECAUDIO, MSGL_DBG2, "Decoded %d -> %d samples\n", in_len,
priv->frame.samples);
return 0;
}
static int decode_audio(sh_audio_t *sh_audio, unsigned char *buf, int minlen,
int maxlen)
static int decode_audio(sh_audio_t *sh, struct mp_audio *buffer, int maxlen)
{
struct priv *priv = sh_audio->context;
AVCodecContext *avctx = priv->avctx;
struct priv *priv = sh->context;
int len = -1;
while (len < minlen) {
if (!priv->output_left) {
if (decode_new_packet(sh_audio) < 0)
break;
continue;
}
if (setup_format(sh_audio, avctx))
return len;
int size = (minlen - len + priv->unitsize - 1);
size -= size % priv->unitsize;
size = FFMIN(size, priv->output_left);
if (size > maxlen)
abort();
memcpy(buf, priv->output, size);
priv->output += size;
priv->output_left -= size;
if (len < 0)
len = size;
else
len += size;
buf += size;
maxlen -= size;
sh_audio->pts_bytes += size;
if (!priv->frame.samples) {
if (decode_new_packet(sh) < 0)
return -1;
}
return len;
if (!mp_audio_config_equals(buffer, &priv->frame))
return 0;
buffer->samples = MPMIN(priv->frame.samples, maxlen);
mp_audio_copy(buffer, 0, &priv->frame, 0, buffer->samples);
mp_audio_skip_samples(&priv->frame, buffer->samples);
sh->pts_offset += buffer->samples;
return 0;
}
static void add_decoders(struct mp_decoder_list *list)

495
audio/decode/ad_mpg123.c
View File

@ -35,7 +35,9 @@
struct ad_mpg123_context {
mpg123_handle *handle;
char new_format;
bool new_format;
int sample_size;
bool need_data;
/* Running mean for bit rate, stream length estimation. */
float mean_rate;
unsigned int mean_count;
@ -45,6 +47,17 @@ struct ad_mpg123_context {
char vbr;
};
static void uninit(sh_audio_t *sh)
{
struct ad_mpg123_context *con = (struct ad_mpg123_context*) sh->context;
mpg123_close(con->handle);
mpg123_delete(con->handle);
talloc_free(sh->context);
sh->context = NULL;
mpg123_exit();
}
/* This initializes libmpg123 and prepares the handle, including funky
* parameters. */
static int preinit(sh_audio_t *sh)
@ -58,7 +71,7 @@ static int preinit(sh_audio_t *sh)
if (mpg123_init() != MPG123_OK)
return 0;
sh->context = malloc(sizeof(struct ad_mpg123_context));
sh->context = talloc_zero(NULL, struct ad_mpg123_context);
con = sh->context;
/* Auto-choice of optimized decoder (first argument NULL). */
con->handle = mpg123_new(NULL, &err);
@ -92,7 +105,12 @@ static int preinit(sh_audio_t *sh)
/* Prevent funky automatic resampling.
* This way, we can be sure that one frame will never produce
* more than 1152 stereo samples. */
* more than 1152 stereo samples.
* Background:
* Going to decode directly to the output buffer. It is important to have
* MPG123_AUTO_RESAMPLE disabled for the buffer size being an all-time
* limit.
* We need at least 1152 samples. dec_audio.c normally guarantees this. */
mpg123_param(con->handle, MPG123_REMOVE_FLAGS, MPG123_AUTO_RESAMPLE, 0.);
return 1;
@ -105,11 +123,125 @@ static int preinit(sh_audio_t *sh)
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 preinit error: %s\n",
mpg123_strerror(con->handle));
if (con->handle)
mpg123_delete(con->handle);
mpg123_exit();
free(sh->context);
sh->context = NULL;
uninit(sh);
return 0;
}
/* libmpg123 has a new format ready; query and store, return return value
of mpg123_getformat() */
static int set_format(sh_audio_t *sh)
{
struct ad_mpg123_context *con = sh->context;
int ret;
long rate;
int channels;
int encoding;
ret = mpg123_getformat(con->handle, &rate, &channels, &encoding);
if (ret == MPG123_OK) {
mp_chmap_from_channels(&sh->channels, channels);
sh->samplerate = rate;
/* Without external force, mpg123 will always choose signed encoding,
* and non-16-bit only on builds that don't support it.
* Be reminded that it doesn't matter to the MPEG file what encoding
* is produced from it. */
switch (encoding) {
case MPG123_ENC_SIGNED_8:
sh->sample_format = AF_FORMAT_S8;
break;
case MPG123_ENC_SIGNED_16:
sh->sample_format = AF_FORMAT_S16_NE;
break;
case MPG123_ENC_SIGNED_32:
sh->sample_format = AF_FORMAT_S32_NE;
break;
case MPG123_ENC_FLOAT_32:
sh->sample_format = AF_FORMAT_FLOAT_NE;
break;
default:
/* This means we got a funny custom build of libmpg123 that only supports an unknown format. */
mp_msg(MSGT_DECAUDIO, MSGL_ERR,
"Bad encoding from mpg123: %i.\n", encoding);
return MPG123_ERR;
}
con->sample_size = channels * (af_fmt2bits(sh->sample_format) / 8);
con->new_format = 0;
}
return ret;
}
static int feed_new_packet(sh_audio_t *sh)
{
struct ad_mpg123_context *con = sh->context;
int ret;
struct demux_packet *pkt = demux_read_packet(sh->gsh);
if (!pkt)
return -1; /* EOF. */
/* Next bytes from that presentation time. */
if (pkt->pts != MP_NOPTS_VALUE) {
sh->pts = pkt->pts;
sh->pts_offset = 0;
}
/* Have to use mpg123_feed() to avoid decoding here. */
ret = mpg123_feed(con->handle, pkt->buffer, pkt->len);
talloc_free(pkt);
if (ret == MPG123_ERR)
return -1;
if (ret == MPG123_NEW_FORMAT)
con->new_format = 1;
return 0;
}
/* Now we really start accessing some data and determining file format.
* Format now is allowed to change on-the-fly. Here is the only point
* that has MPlayer react to errors. We have to pray that exceptional
* erros in other places simply cannot occur. */
static int init(sh_audio_t *sh, const char *decoder)
{
struct ad_mpg123_context *con = sh->context;
int ret;
ret = mpg123_open_feed(con->handle);
if (ret != MPG123_OK)
goto fail;
for (int n = 0; ; n++) {
if (feed_new_packet(sh) < 0) {
ret = MPG123_NEED_MORE;
goto fail;
}
size_t got_now = 0;
ret = mpg123_decode_frame(con->handle, NULL, NULL, &got_now);
if (ret == MPG123_OK || ret == MPG123_NEW_FORMAT) {
ret = set_format(sh);
if (ret == MPG123_OK)
break;
}
if (ret != MPG123_NEED_MORE)
goto fail;
// max. 16 retries (randomly chosen number)
if (n > 16) {
ret = MPG123_NEED_MORE;
goto fail;
}
}
return 1;
fail:
if (ret == MPG123_NEED_MORE) {
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "Could not find mp3 stream.\n");
} else {
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 init error: %s\n",
mpg123_strerror(con->handle));
}
uninit(sh);
return 0;
}
@ -126,274 +258,17 @@ static int compute_bitrate(struct mpg123_frameinfo *i)
samples_per_frame[i->version][i->layer] + 0.5);
}
/* Opted against the header printout from old mp3lib, too much
* irrelevant info. This is modelled after the mpg123 app's
* standard output line.
* If more verbosity is demanded, one can add more detail and
* also throw in ID3v2 info which libmpg123 collects anyway. */
static void print_header_compact(struct mpg123_frameinfo *i)
{
static const char *smodes[5] = {
"stereo", "joint-stereo", "dual-channel", "mono", "invalid"
};
static const char *layers[4] = {
"Unknown", "I", "II", "III"
};
static const char *versions[4] = {
"1.0", "2.0", "2.5", "x.x"
};
mp_msg(MSGT_DECAUDIO, MSGL_V, "MPEG %s layer %s, ",
versions[i->version], layers[i->layer]);
switch (i->vbr) {
case MPG123_CBR:
if (i->bitrate)
mp_msg(MSGT_DECAUDIO, MSGL_V, "%d kbit/s", i->bitrate);
else
mp_msg(MSGT_DECAUDIO, MSGL_V, "%d kbit/s (free format)",
compute_bitrate(i));
break;
case MPG123_VBR:
mp_msg(MSGT_DECAUDIO, MSGL_V, "VBR");
break;
case MPG123_ABR:
mp_msg(MSGT_DECAUDIO, MSGL_V, "%d kbit/s ABR", i->abr_rate);
break;
default:
mp_msg(MSGT_DECAUDIO, MSGL_V, "???");
}
mp_msg(MSGT_DECAUDIO, MSGL_V, ", %ld Hz %s\n", i->rate,
smodes[i->mode]);
}
/* libmpg123 has a new format ready; query and store, return return value
of mpg123_getformat() */
static int set_format(sh_audio_t *sh, struct ad_mpg123_context *con)
{
int ret;
long rate;
int channels;
int encoding;
ret = mpg123_getformat(con->handle, &rate, &channels, &encoding);
if(ret == MPG123_OK) {
mp_chmap_from_channels(&sh->channels, channels);
sh->samplerate = rate;
/* Without external force, mpg123 will always choose signed encoding,
* and non-16-bit only on builds that don't support it.
* Be reminded that it doesn't matter to the MPEG file what encoding
* is produced from it. */
switch (encoding) {
case MPG123_ENC_SIGNED_8:
sh->sample_format = AF_FORMAT_S8;
break;
case MPG123_ENC_SIGNED_16:
sh->sample_format = AF_FORMAT_S16_NE;
break;
/* To stay compatible with the oldest libmpg123 headers, do not rely
* on float and 32 bit encoding symbols being defined.
* Those formats came later */
case 0x1180: /* MPG123_ENC_SIGNED_32 */
sh->sample_format = AF_FORMAT_S32_NE;
break;
case 0x200: /* MPG123_ENC_FLOAT_32 */
sh->sample_format = AF_FORMAT_FLOAT_NE;
break;
default:
/* This means we got a funny custom build of libmpg123 that only supports an unknown format. */
mp_msg(MSGT_DECAUDIO, MSGL_ERR,
"Bad encoding from mpg123: %i.\n", encoding);
return MPG123_ERR;
}
/* Going to decode directly to MPlayer's memory. It is important
* to have MPG123_AUTO_RESAMPLE disabled for the buffer size
* being an all-time limit. */
sh->audio_out_minsize = 1152 * 2 * (af_fmt2bits(sh->sample_format) / 8);
con->new_format = 0;
}
return ret;
}
/* This tries to extract a requested amount of decoded data.
* Even when you request 0 bytes, it will feed enough input so that
* the decoder _could_ have delivered something.
* Returns byte count >= 0, -1 on error.
*
* Thoughts on exact pts keeping:
* We have to assume that MPEG frames are cut in pieces by packet boundaries.
* Also, it might be possible that the first packet does not contain enough
* data to ensure initial stream sync... or re-sync on erroneous streams.
* So we need something robust to relate the decoded byte count to the correct
* time stamp. This is tricky, though. From the outside, you cannot tell if,
* after having fed two packets until the first output arrives, one should
* start counting from the first packet's pts or the second packet's.
* So, let's just count from the last fed package's pts. If the packets are
* exactly cut to MPEG frames, this will cause one frame mismatch in the
* beginning (when mpg123 peeks ahead for the following header), but will
* be corrected with the third frame already. One might add special code to
* not increment the base pts past the first packet's after a resync before
* the first decoded bytes arrived. */
static int decode_a_bit(sh_audio_t *sh, unsigned char *buf, int count)
{
int ret = MPG123_OK;
int got = 0;
struct ad_mpg123_context *con = sh->context;
/* There will be one MPG123_NEW_FORMAT message on first open.
* This will be handled in init(). */
do {
size_t got_now = 0;
/* Fetch new format now, after old data has been used. */
if(con->new_format)
ret = set_format(sh, con);
/* Feed the decoder. This will only fire from the second round on. */
if (ret == MPG123_NEED_MORE) {
/* Feed more input data. */
struct demux_packet *pkt = demux_read_packet(sh->gsh);
if (!pkt)
break; /* Apparently that's it. EOF. */
/* Next bytes from that presentation time. */
if (pkt->pts != MP_NOPTS_VALUE) {
sh->pts = pkt->pts;
sh->pts_bytes = 0;
}
/* Have to use mpg123_feed() to avoid decoding here. */
ret = mpg123_feed(con->handle, pkt->buffer, pkt->len);
talloc_free(pkt);
if (ret == MPG123_ERR)
break;
/* Indication of format change is possible here (from mpg123_decode()). */
if(ret == MPG123_NEW_FORMAT) {
con->new_format = 1;
if(got)
break; /* Do not switch format during a chunk. */
ret = set_format(sh, con);
}
}
/* Theoretically, mpg123 could return MPG123_DONE, so be prepared.
* Should not happen in our usage, but it is a valid return code. */
else if (ret == MPG123_ERR || ret == MPG123_DONE)
break;
/* Try to decode a bit. This is the return value that counts
* for the loop condition. */
if (!buf) { /* fake call just for feeding to get format */
ret = set_format(sh, con);
} else { /* This is the decoding. One frame at a time. */
ret = mpg123_replace_buffer(con->handle, buf, count);
if (ret == MPG123_OK)
ret = mpg123_decode_frame(con->handle, NULL, NULL, &got_now);
}
got += got_now;
sh->pts_bytes += got_now;
/* Indication of format change should happen here. */
if(ret == MPG123_NEW_FORMAT) {
con->new_format = 1;
if(got)
break; /* Do not switch format during a chunk. */
ret = set_format(sh, con);
}
} while (ret == MPG123_NEED_MORE || (got == 0 && count != 0));
if (ret == MPG123_ERR) {
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 decoding failed: %s\n",
mpg123_strerror(con->handle));
}
return got;
}
/* Close, reopen stream. Feed data until we know the format of the stream.
* 1 on success, 0 on error */
static int reopen_stream(sh_audio_t *sh)
{
struct ad_mpg123_context *con = (struct ad_mpg123_context*) sh->context;
mpg123_close(con->handle);
/* No resetting of the context:
* We do not want to loose the mean bitrate data. */
/* Open and make sure we have fed enough data to get stream properties. */
if (MPG123_OK == mpg123_open_feed(con->handle) &&
/* Feed data until mpg123 is ready (has found stream beginning). */
!decode_a_bit(sh, NULL, 0) &&
set_format(sh, con) == MPG123_OK) { /* format setting again just for return value */
return 1;
} else {
mp_msg(MSGT_DECAUDIO, MSGL_ERR,
"mpg123 failed to reopen stream: %s\n",
mpg123_strerror(con->handle));
return 0;
}
}
/* Now we really start accessing some data and determining file format.
* Format now is allowed to change on-the-fly. Here is the only point
* that has MPlayer react to errors. We have to pray that exceptional
* erros in other places simply cannot occur. */
static int init(sh_audio_t *sh, const char *decoder)
{
mpg123_id3v2 *v2;
struct mpg123_frameinfo finfo;
struct ad_mpg123_context *con = sh->context;
con->new_format = 0;
if (reopen_stream(sh) &&
/* Get MPEG header info. */
MPG123_OK == mpg123_info(con->handle, &finfo) &&
/* Since we queried format, mpg123 should have read past ID3v2 tags.
* We need to decide if printing of UTF-8 encoded text info is wanted. */
MPG123_OK == mpg123_id3(con->handle, NULL, &v2)) {
/* If we are here, we passed all hurdles. Yay! Extract the info. */
print_header_compact(&finfo);
/* Do we want to print out the UTF-8 Id3v2 info?
if (v2)
print_id3v2(v2); */
/* Have kb/s, want B/s
* For VBR, the first frame will be a bad estimate. */
sh->i_bps = (finfo.bitrate ? finfo.bitrate : compute_bitrate(&finfo))
* 1000 / 8;
con->delay = 1;
con->mean_rate = 0.;
con->mean_count = 0;
con->vbr = (finfo.vbr != MPG123_CBR);
return 1;
} else {
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 init error: %s\n",
mpg123_strerror(con->handle));
return 0;
}
}
static void uninit(sh_audio_t *sh)
{
struct ad_mpg123_context *con = (struct ad_mpg123_context*) sh->context;
mpg123_close(con->handle);
mpg123_delete(con->handle);
free(sh->context);
sh->context = NULL;
mpg123_exit();
}
/* Update mean bitrate. This could be dropped if accurate time display
* on audio file playback is not desired. */
static void update_info(sh_audio_t *sh)
{
struct ad_mpg123_context *con = sh->context;
if (con->vbr && --con->delay < 1) {
struct mpg123_frameinfo finfo;
if (MPG123_OK == mpg123_info(con->handle, &finfo)) {
struct mpg123_frameinfo finfo;
if (mpg123_info(con->handle, &finfo) != MPG123_OK)
return;
if (finfo.vbr != MPG123_CBR) {
if (--con->delay < 1) {
if (++con->mean_count > ((unsigned int) -1) / 2)
con->mean_count = ((unsigned int) -1) / 4;
@ -404,42 +279,80 @@ static void update_info(sh_audio_t *sh)
con->delay = 10;
}
} else {
sh->i_bps = (finfo.bitrate ? finfo.bitrate : compute_bitrate(&finfo))
* 1000 / 8;
con->delay = 1;
con->mean_rate = 0.;
con->mean_count = 0;
}
}
static int decode_audio(sh_audio_t *sh, unsigned char *buf, int minlen,
int maxlen)
static int decode_audio(sh_audio_t *sh, struct mp_audio *buffer, int maxlen)
{
int bytes;
struct ad_mpg123_context *con = sh->context;
void *buf = buffer->planes[0];
int ret;
bytes = decode_a_bit(sh, buf, maxlen);
/* This EOF is ignored, apparently, until input data is exhausted. */
if (bytes == 0)
return -1; /* EOF */
if (con->new_format) {
ret = set_format(sh);
if (ret == MPG123_OK) {
return 0; // let caller handle format change
} else if (ret == MPG123_NEED_MORE) {
con->need_data = true;
} else {
goto mpg123_fail;
}
}
if (con->need_data) {
if (feed_new_packet(sh) < 0)
return -1;
}
size_t got_now = 0;
ret = mpg123_replace_buffer(con->handle, buf, maxlen * con->sample_size);
if (ret != MPG123_OK)
goto mpg123_fail;
ret = mpg123_decode_frame(con->handle, NULL, NULL, &got_now);
int got_samples = got_now / con->sample_size;
buffer->samples += got_samples;
sh->pts_offset += got_samples;
if (ret == MPG123_NEW_FORMAT) {
con->new_format = true;
} else if (ret == MPG123_NEED_MORE) {
con->need_data = true;
} else if (ret != MPG123_OK && ret != MPG123_DONE) {
goto mpg123_fail;
}
update_info(sh);
return bytes;
return 0;
mpg123_fail:
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 decoding error: %s\n",
mpg123_strerror(con->handle));
return -1;
}
static int control(sh_audio_t *sh, int cmd, void *arg)
{
struct ad_mpg123_context *con = sh->context;
switch (cmd) {
case ADCTRL_RESYNC_STREAM:
/* Close/reopen the stream for mpg123 to make sure it doesn't
* think that it still knows the exact stream position.
* Otherwise, we would have funny effects from the gapless code.
* Oh, and it helps to minimize artifacts from jumping in the stream. */
if (reopen_stream(sh)) {
update_info(sh);
return CONTROL_TRUE;
} else {
/* MPlayer ignores this case! It just keeps on decoding.
* So we have to make sure resync never fails ... */
mpg123_close(con->handle);
if (mpg123_open_feed(con->handle) != MPG123_OK) {
mp_msg(MSGT_DECAUDIO, MSGL_ERR,
"mpg123 cannot reopen stream for resync.\n");
"mpg123 failed to reopen stream: %s\n",
mpg123_strerror(con->handle));
return CONTROL_FALSE;
}
break;
return CONTROL_TRUE;
}
return CONTROL_UNKNOWN;
}

55
audio/decode/ad_spdif.c
View File

@ -19,6 +19,7 @@
*/
#include <string.h>
#include <assert.h>
#include <libavformat/avformat.h>
#include <libavcodec/avcodec.h>
@ -184,37 +185,43 @@ fail:
return 0;
}
static int decode_audio(sh_audio_t *sh, unsigned char *buf,
int minlen, int maxlen)
static int decode_audio(sh_audio_t *sh, struct mp_audio *buffer, int maxlen)
{
struct spdifContext *spdif_ctx = sh->context;
AVFormatContext *lavf_ctx = spdif_ctx->lavf_ctx;
int sstride = 2 * sh->channels.num;
assert(sstride == buffer->sstride);
if (maxlen < spdif_ctx->iec61937_packet_size)
return 0;
spdif_ctx->out_buffer_len = 0;
spdif_ctx->out_buffer_size = maxlen;
spdif_ctx->out_buffer = buf;
while (spdif_ctx->out_buffer_len + spdif_ctx->iec61937_packet_size < maxlen
&& spdif_ctx->out_buffer_len < minlen) {
struct demux_packet *mpkt = demux_read_packet(sh->gsh);
if (!mpkt)
break;
AVPacket pkt;
mp_set_av_packet(&pkt, mpkt);
pkt.pts = pkt.dts = 0;
mp_msg(MSGT_DECAUDIO, MSGL_V, "spdif packet, size=%d\n", pkt.size);
if (mpkt->pts != MP_NOPTS_VALUE) {
sh->pts = mpkt->pts;
sh->pts_bytes = 0;
}
int out_len = spdif_ctx->out_buffer_len;
int ret = av_write_frame(lavf_ctx, &pkt);
avio_flush(lavf_ctx->pb);
sh->pts_bytes += spdif_ctx->out_buffer_len - out_len;
talloc_free(mpkt);
if (ret < 0)
break;
spdif_ctx->out_buffer = buffer->planes[0];
struct demux_packet *mpkt = demux_read_packet(sh->gsh);
if (!mpkt)
return 0;
AVPacket pkt;
mp_set_av_packet(&pkt, mpkt);
pkt.pts = pkt.dts = 0;
mp_msg(MSGT_DECAUDIO, MSGL_V, "spdif packet, size=%d\n", pkt.size);
if (mpkt->pts != MP_NOPTS_VALUE) {
sh->pts = mpkt->pts;
sh->pts_offset = 0;
}
return spdif_ctx->out_buffer_len;
int out_len = spdif_ctx->out_buffer_len;
int ret = av_write_frame(lavf_ctx, &pkt);
avio_flush(lavf_ctx->pb);
sh->pts_offset += (spdif_ctx->out_buffer_len - out_len) / sstride;
talloc_free(mpkt);
if (ret < 0)
return -1;
buffer->samples = spdif_ctx->out_buffer_len / sstride;
return 0;
}
static int control(sh_audio_t *sh, int cmd, void *arg)

149
audio/decode/dec_audio.c
View File

@ -38,6 +38,7 @@
#include "dec_audio.h"
#include "ad.h"
#include "audio/format.h"
#include "audio/audio.h"
#include "audio/audio_buffer.h"
#include "audio/filter/af.h"
@ -55,31 +56,29 @@ static const struct ad_functions * const ad_drivers[] = {
NULL
};
// At least ad_mpg123 needs to be able to decode this many samples at once
#define DECODE_MAX_UNIT 1152
// At least 8192 samples, plus hack for ad_mpg123
#define DECODE_BUFFER_SAMPLES (8192 + DECODE_MAX_UNIT)
// Drop audio buffer and reinit it (after format change)
static void reinit_audio_buffer(sh_audio_t *sh)
{
mp_audio_buffer_reinit_fmt(sh->decode_buffer, sh->sample_format,
&sh->channels, sh->samplerate);
mp_audio_buffer_preallocate_min(sh->decode_buffer, DECODE_BUFFER_SAMPLES);
}
static int init_audio_codec(sh_audio_t *sh_audio, const char *decoder)
{
assert(!sh_audio->initialized);
resync_audio_stream(sh_audio);
sh_audio->sample_format = AF_FORMAT_FLOAT_NE;
sh_audio->audio_out_minsize = 8192; // default, preinit() may change it
if (!sh_audio->ad_driver->preinit(sh_audio)) {
mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Audio decoder preinit failed.\n");
return 0;
}
const int base_size = 65536;
// At least 64 KiB plus rounding up to next decodable unit size
sh_audio->a_buffer_size = base_size + sh_audio->audio_out_minsize;
mp_tmsg(MSGT_DECAUDIO, MSGL_V,
"dec_audio: Allocating %d + %d = %d bytes for output buffer.\n",
sh_audio->audio_out_minsize, base_size,
sh_audio->a_buffer_size);
sh_audio->a_buffer = av_mallocz(sh_audio->a_buffer_size);
if (!sh_audio->a_buffer)
abort();
sh_audio->a_buffer_len = 0;
if (!sh_audio->ad_driver->init(sh_audio, decoder)) {
mp_tmsg(MSGT_DECAUDIO, MSGL_V, "Audio decoder init failed.\n");
uninit_audio(sh_audio); // free buffers
@ -88,13 +87,18 @@ static int init_audio_codec(sh_audio_t *sh_audio, const char *decoder)
sh_audio->initialized = 1;
if (mp_chmap_is_empty(&sh_audio->channels) || !sh_audio->samplerate) {
if (mp_chmap_is_empty(&sh_audio->channels) || !sh_audio->samplerate ||
!sh_audio->sample_format)
{
mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Audio decoder did not specify "
"audio format!\n");
uninit_audio(sh_audio); // free buffers
return 0;
}
sh_audio->decode_buffer = mp_audio_buffer_create(NULL);
reinit_audio_buffer(sh_audio);
return 1;
}
@ -188,7 +192,8 @@ void uninit_audio(sh_audio_t *sh_audio)
}
talloc_free(sh_audio->gsh->decoder_desc);
sh_audio->gsh->decoder_desc = NULL;
av_freep(&sh_audio->a_buffer);
talloc_free(sh_audio->decode_buffer);
sh_audio->decode_buffer = NULL;
}
@ -235,37 +240,44 @@ int init_audio_filters(sh_audio_t *sh_audio, int in_samplerate,
static int filter_n_bytes(sh_audio_t *sh, struct mp_audio_buffer *outbuf,
int len)
{
assert(len - 1 + sh->audio_out_minsize <= sh->a_buffer_size);
int error = 0;
// Decode more bytes if needed
int old_samplerate = sh->samplerate;
struct mp_chmap old_channels = sh->channels;
int old_sample_format = sh->sample_format;
while (sh->a_buffer_len < len) {
unsigned char *buf = sh->a_buffer + sh->a_buffer_len;
int minlen = len - sh->a_buffer_len;
int maxlen = sh->a_buffer_size - sh->a_buffer_len;
int ret = sh->ad_driver->decode_audio(sh, buf, minlen, maxlen);
int format_change = sh->samplerate != old_samplerate
|| !mp_chmap_equals(&sh->channels, &old_channels)
|| sh->sample_format != old_sample_format;
if (ret <= 0 || format_change) {
error = format_change ? -2 : -1;
// samples from format-changing call get discarded too
len = sh->a_buffer_len;
struct mp_audio config;
mp_audio_buffer_get_format(sh->decode_buffer, &config);
while (mp_audio_buffer_samples(sh->decode_buffer) < len) {
int maxlen = mp_audio_buffer_get_write_available(sh->decode_buffer);
if (maxlen < DECODE_MAX_UNIT)
break;
struct mp_audio buffer;
mp_audio_buffer_get_write_buffer(sh->decode_buffer, maxlen, &buffer);
buffer.samples = 0;
error = sh->ad_driver->decode_audio(sh, &buffer, maxlen);
if (error < 0)
break;
// Commit the data just read as valid data
mp_audio_buffer_finish_write(sh->decode_buffer, buffer.samples);
// Format change
if (sh->samplerate != config.rate ||
!mp_chmap_equals(&sh->channels, &config.channels) ||
sh->sample_format != config.format)
{
// If there are still samples left in the buffer, let them drain
// first, and don't signal a format change to the caller yet.
if (mp_audio_buffer_samples(sh->decode_buffer) > 0)
break;
reinit_audio_buffer(sh);
error = -2;
break;
}
sh->a_buffer_len += ret;
}
// Filter
struct mp_audio filter_input = {
.planes = {sh->a_buffer},
};
mp_audio_copy_config(&filter_input, &sh->afilter->input);
filter_input.samples = len / filter_input.sstride;
struct mp_audio filter_input;
mp_audio_buffer_peek(sh->decode_buffer, &filter_input);
filter_input.rate = sh->afilter->input.rate; // due to playback speed change
len = MPMIN(filter_input.samples, len);
filter_input.samples = len;
struct mp_audio *filter_output = af_play(sh->afilter, &filter_input);
if (!filter_output)
@ -273,8 +285,7 @@ static int filter_n_bytes(sh_audio_t *sh, struct mp_audio_buffer *outbuf,
mp_audio_buffer_append(outbuf, filter_output);
// remove processed data from decoder buffer:
sh->a_buffer_len -= len;
memmove(sh->a_buffer, sh->a_buffer + len, sh->a_buffer_len);
mp_audio_buffer_skip(sh->decode_buffer, len);
return error;
}
@ -289,33 +300,27 @@ int decode_audio(sh_audio_t *sh_audio, struct mp_audio_buffer *outbuf,
{
// Indicates that a filter seems to be buffering large amounts of data
int huge_filter_buffer = 0;
int sstride =
af_fmt2bits(sh_audio->sample_format) / 8 * sh_audio->channels.num;
// Decoded audio must be cut at boundaries of this many bytes
int unitsize = sstride * 16;
// Decoded audio must be cut at boundaries of this many samples
// (Note: the reason for this is unknown, possibly a refactoring artifact)
int unitsize = 16;
/* Filter output size will be about filter_multiplier times input size.
* If some filter buffers audio in big blocks this might only hold
* as average over time. */
double filter_multiplier = af_calc_filter_multiplier(sh_audio->afilter);
/* If the decoder set audio_out_minsize then it can do the equivalent of
* "while (output_len < target_len) output_len += audio_out_minsize;",
* so we must guarantee there is at least audio_out_minsize-1 bytes
* more space in the output buffer than the minimum length we try to
* decode. */
int max_decode_len = sh_audio->a_buffer_size - sh_audio->audio_out_minsize;
if (!unitsize)
return -1;
max_decode_len -= max_decode_len % unitsize;
int prev_buffered = -1;
while (minsamples >= 0) {
int buffered = mp_audio_buffer_samples(outbuf);
if (minsamples < buffered || buffered == prev_buffered)
break;
prev_buffered = buffered;
while (minsamples >= 0 && mp_audio_buffer_samples(outbuf) < minsamples) {
int decsamples = (minsamples - mp_audio_buffer_samples(outbuf))
/ filter_multiplier;
int declen = decsamples * sstride;
int decsamples = (minsamples - buffered) / filter_multiplier;
// + some extra for possible filter buffering
declen += unitsize << 5;
if (huge_filter_buffer)
decsamples += 1 << unitsize;
if (huge_filter_buffer) {
/* Some filter must be doing significant buffering if the estimated
* input length didn't produce enough output from filters.
* Feed the filters 2k bytes at a time until we have enough output.
@ -324,15 +329,14 @@ int decode_audio(sh_audio_t *sh_audio, struct mp_audio_buffer *outbuf,
* to get audio data and buffer video frames in memory while doing
* so. However the performance impact of either is probably not too
* significant as long as the value is not completely insane. */
declen = 2000;
declen -= declen % unitsize;
if (declen > max_decode_len)
declen = max_decode_len;
else
/* if this iteration does not fill buffer, we must have lots
* of buffering in filters */
huge_filter_buffer = 1;
int res = filter_n_bytes(sh_audio, outbuf, declen);
decsamples = 2000;
}
/* if this iteration does not fill buffer, we must have lots
* of buffering in filters */
huge_filter_buffer = 1;
int res = filter_n_bytes(sh_audio, outbuf, decsamples);
if (res < 0)
return res;
}
@ -342,6 +346,7 @@ int decode_audio(sh_audio_t *sh_audio, struct mp_audio_buffer *outbuf,
void resync_audio_stream(sh_audio_t *sh_audio)
{
sh_audio->pts = MP_NOPTS_VALUE;
sh_audio->pts_offset = 0;
if (!sh_audio->initialized)
return;
sh_audio->ad_driver->control(sh_audio, ADCTRL_RESYNC_STREAM, NULL);

32
audio/reorder_ch.c
View File

@ -63,38 +63,6 @@ void reorder_to_planar(void *restrict out, const void *restrict in,
reorder_to_planar_(out, in, size, nchan, nmemb);
}
static inline void reorder_to_packed_(uint8_t *out, uint8_t **in,
size_t size, size_t nchan, size_t nmemb)
{
size_t outstep = nchan * size;
for (size_t c = 0; c < nchan; ++c) {
char *outptr = out + c * size;
char *inptr = in[c];
for (size_t i = 0; i < nmemb; ++i, outptr += outstep, inptr += size) {
memcpy(outptr, inptr, size);
}
}
}
// out = destination array of packed samples of given size, nmemb frames
// in[channel] = source array of samples for the given channel
void reorder_to_packed(uint8_t *out, uint8_t **in,
size_t size, size_t nchan, size_t nmemb)
{
if (nchan == 1)
memcpy(out, in, size * nchan * nmemb);
// See reorder_to_planar() why this is done this way
else if (size == 1)
reorder_to_packed_(out, in, 1, nchan, nmemb);
else if (size == 2)
reorder_to_packed_(out, in, 2, nchan, nmemb);
else if (size == 4)
reorder_to_packed_(out, in, 4, nchan, nmemb);
else
reorder_to_packed_(out, in, size, nchan, nmemb);
}
#define MAX_SAMPLESIZE 8
static void reorder_channels_(uint8_t *restrict data, int *restrict ch_order,

2
audio/reorder_ch.h
View File

@ -27,8 +27,6 @@
void reorder_to_planar(void *restrict out, const void *restrict in,
size_t size, size_t nchan, size_t nmemb);
void reorder_to_packed(uint8_t *out, uint8_t **in,
size_t size, size_t nchan, size_t nmemb);
void reorder_channels(void *restrict data, int *restrict ch_order,
size_t sample_size, size_t num_ch, size_t num_frames);

1
demux/demux.c
View File

@ -259,7 +259,6 @@ struct sh_stream *new_sh_stream(demuxer_t *demuxer, enum stream_type type)
struct sh_audio *sht = talloc_zero(demuxer, struct sh_audio);
sht->gsh = sh;
sht->opts = sh->opts;
sht->sample_format = AF_FORMAT_S16_NE;
sh->audio = sht;
break;
}

9
demux/stheader.h
View File

@ -92,11 +92,8 @@ typedef struct sh_audio {
int samplerate;
struct mp_chmap channels;
int i_bps; // == bitrate (compressed bytes/sec)
// decoder buffers:
int audio_out_minsize; // minimal output from decoder may be this much
char *a_buffer; // buffer for decoder output
int a_buffer_len;
int a_buffer_size;
// decoder state:
struct mp_audio_buffer *decode_buffer;
struct af_stream *afilter; // the audio filter stream
const struct ad_functions *ad_driver;
// win32-compatible codec parameters:
@ -104,7 +101,7 @@ typedef struct sh_audio {
// note codec extradata may be either under "wf" or "codecdata"
unsigned char *codecdata;
int codecdata_len;
int pts_bytes; // bytes output by decoder after last known pts
int pts_offset; // number of samples output by decoder after last known pts
} sh_audio_t;
typedef struct sh_video {

13
mpvcore/player/audio.c
View File

@ -180,12 +180,9 @@ no_audio:
double written_audio_pts(struct MPContext *mpctx)
{
sh_audio_t *sh_audio = mpctx->sh_audio;
if (!sh_audio)
if (!sh_audio || !sh_audio->initialized)
return MP_NOPTS_VALUE;
double bps = sh_audio->channels.num * sh_audio->samplerate *
(af_fmt2bits(sh_audio->sample_format) / 8);
// first calculate the end pts of audio that has been output by decoder
double a_pts = sh_audio->pts;
if (a_pts == MP_NOPTS_VALUE)
@ -194,13 +191,13 @@ double written_audio_pts(struct MPContext *mpctx)
// sh_audio->pts is the timestamp of the latest input packet with
// known pts that the decoder has decoded. sh_audio->pts_bytes is
// the amount of bytes the decoder has written after that timestamp.
a_pts += sh_audio->pts_bytes / bps;
a_pts += sh_audio->pts_offset / (double)sh_audio->samplerate;
// Now a_pts hopefully holds the pts for end of audio from decoder.
// Subtract data in buffers between decoder and audio out.
// Decoded but not filtered
a_pts -= sh_audio->a_buffer_len / bps;
a_pts -= mp_audio_buffer_seconds(sh_audio->decode_buffer);
// Data buffered in audio filters, measured in seconds of "missing" output
double buffered_output = af_calc_delay(sh_audio->afilter);
@ -446,6 +443,6 @@ void clear_audio_output_buffers(struct MPContext *mpctx)
// Drop decoded data queued for filtering.
void clear_audio_decode_buffers(struct MPContext *mpctx)
{
if (mpctx->sh_audio)
mpctx->sh_audio->a_buffer_len = 0;
if (mpctx->sh_audio && mpctx->sh_audio->decode_buffer)
mp_audio_buffer_clear(mpctx->sh_audio->decode_buffer);
}