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mpv/audio/out/ao_lavc.c
wm4 380fc765e4 audio/out: prepare for non-interleaved audio 
This comes with two internal AO API changes:

1. ao_driver.play now can take non-interleaved audio. For this purpose,
the data pointer is changed to void **data, where data[0] corresponds to
the pointer in the old API. Also, the len argument as well as the return
value are now in samples, not bytes. "Sample" in this context means the
unit of the smallest possible audio frame, i.e. sample_size * channels.

2. ao_driver.get_space now returns samples instead of bytes. (Similar to
the play function.)

Change all AOs to use the new API.

The AO API as exposed to the rest of the player still uses the old API.
It's emulated in ao.c. This is purely to split the commits changing all
AOs and the commits adding actual support for outputting N-I audio.
2013-11-12 23:27:51 +01:00

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/*
* audio encoding using libavformat
* Copyright (C) 2011-2012 Rudolf Polzer <divVerent@xonotic.org>
* NOTE: this file is partially based on ao_pcm.c by Atmosfear
*
* This file is part of mpv.
*
* MPlayer is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* MPlayer 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with MPlayer; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include <stdio.h>
#include <stdlib.h>
#include <libavutil/common.h>
#include <libavutil/audioconvert.h>
#include "compat/libav.h"
#include "config.h"
#include "mpvcore/options.h"
#include "mpvcore/mp_common.h"
#include "audio/format.h"
#include "audio/reorder_ch.h"
#include "talloc.h"
#include "ao.h"
#include "mpvcore/mp_msg.h"
#include "mpvcore/encode_lavc.h"
static const char *sample_padding_signed = "\x00\x00\x00\x00";
static const char *sample_padding_u8 = "\x80";
static const char *sample_padding_float = "\x00\x00\x00\x00";
struct priv {
uint8_t *buffer;
size_t buffer_size;
AVStream *stream;
bool planarize;
int pcmhack;
int aframesize;
int aframecount;
int offset;
int offset_left;
int64_t savepts;
int framecount;
int64_t lastpts;
int sample_size;
const void *sample_padding;
double expected_next_pts;
AVRational worst_time_base;
int worst_time_base_is_stream;
};
// open & setup audio device
static int init(struct ao *ao)
{
struct priv *ac = talloc_zero(ao, struct priv);
const enum AVSampleFormat *sampleformat;
AVCodec *codec;
if (!encode_lavc_available(ao->encode_lavc_ctx)) {
MP_ERR(ao, "the option --o (output file) must be specified\n");
return -1;
}
ac->stream = encode_lavc_alloc_stream(ao->encode_lavc_ctx,
AVMEDIA_TYPE_AUDIO);
if (!ac->stream) {
MP_ERR(ao, "could not get a new audio stream\n");
return -1;
}
codec = encode_lavc_get_codec(ao->encode_lavc_ctx, ac->stream);
// ac->stream->time_base.num = 1;
// ac->stream->time_base.den = ao->samplerate;
// doing this breaks mpeg2ts in ffmpeg
// which doesn't properly force the time base to be 90000
// furthermore, ffmpeg.c doesn't do this either and works
ac->stream->codec->time_base.num = 1;
ac->stream->codec->time_base.den = ao->samplerate;
ac->stream->codec->sample_rate = ao->samplerate;
struct mp_chmap_sel sel = {0};
mp_chmap_sel_add_any(&sel);
if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
return -1;
mp_chmap_reorder_to_lavc(&ao->channels);
ac->stream->codec->channels = ao->channels.num;
ac->stream->codec->channel_layout = mp_chmap_to_lavc(&ao->channels);
ac->stream->codec->sample_fmt = AV_SAMPLE_FMT_NONE;
ao->format = af_fmt_from_planar(ao->format);
{
// first check if the selected format is somewhere in the list of
// supported formats by the codec
for (sampleformat = codec->sample_fmts;
sampleformat && *sampleformat != AV_SAMPLE_FMT_NONE;
++sampleformat) {
switch (*sampleformat) {
case AV_SAMPLE_FMT_U8:
case AV_SAMPLE_FMT_U8P:
if (ao->format == AF_FORMAT_U8)
goto out_search;
break;
case AV_SAMPLE_FMT_S16:
case AV_SAMPLE_FMT_S16P:
if (ao->format == AF_FORMAT_S16_BE)
goto out_search;
if (ao->format == AF_FORMAT_S16_LE)
goto out_search;
break;
case AV_SAMPLE_FMT_S32:
case AV_SAMPLE_FMT_S32P:
if (ao->format == AF_FORMAT_S32_BE)
goto out_search;
if (ao->format == AF_FORMAT_S32_LE)
goto out_search;
break;
case AV_SAMPLE_FMT_FLT:
case AV_SAMPLE_FMT_FLTP:
if (ao->format == AF_FORMAT_FLOAT_BE)
goto out_search;
if (ao->format == AF_FORMAT_FLOAT_LE)
goto out_search;
break;
// FIXME do we need support for AV_SAMPLE_FORMAT_DBL/DBLP?
default:
break;
}
}
out_search:
;
}
if (!sampleformat || *sampleformat == AV_SAMPLE_FMT_NONE) {
// if the selected format is not supported, we have to pick the first
// one we CAN support
// note: not needing to select endianness here, as the switch() below
// does that anyway for us
for (sampleformat = codec->sample_fmts;
sampleformat && *sampleformat != AV_SAMPLE_FMT_NONE;
++sampleformat) {
switch (*sampleformat) {
case AV_SAMPLE_FMT_U8:
case AV_SAMPLE_FMT_U8P:
ao->format = AF_FORMAT_U8;
goto out_takefirst;
case AV_SAMPLE_FMT_S16:
case AV_SAMPLE_FMT_S16P:
ao->format = AF_FORMAT_S16_NE;
goto out_takefirst;
case AV_SAMPLE_FMT_S32:
case AV_SAMPLE_FMT_S32P:
ao->format = AF_FORMAT_S32_NE;
goto out_takefirst;
case AV_SAMPLE_FMT_FLT:
case AV_SAMPLE_FMT_FLTP:
ao->format = AF_FORMAT_FLOAT_NE;
goto out_takefirst;
// FIXME do we need support for AV_SAMPLE_FORMAT_DBL/DBLP?
default:
break;
}
}
out_takefirst:
;
}
switch (ao->format) {
// now that we have chosen a format, set up the fields for it, boldly
// switching endianness if needed (mplayer code will convert for us
// anyway, but ffmpeg always expects native endianness)
case AF_FORMAT_U8:
ac->stream->codec->sample_fmt = AV_SAMPLE_FMT_U8;
ac->sample_size = 1;
ac->sample_padding = sample_padding_u8;
ao->format = AF_FORMAT_U8;
break;
default:
case AF_FORMAT_S16_BE:
case AF_FORMAT_S16_LE:
ac->stream->codec->sample_fmt = AV_SAMPLE_FMT_S16;
ac->sample_size = 2;
ac->sample_padding = sample_padding_signed;
ao->format = AF_FORMAT_S16_NE;
break;
case AF_FORMAT_S32_BE:
case AF_FORMAT_S32_LE:
ac->stream->codec->sample_fmt = AV_SAMPLE_FMT_S32;
ac->sample_size = 4;
ac->sample_padding = sample_padding_signed;
ao->format = AF_FORMAT_S32_NE;
break;
case AF_FORMAT_FLOAT_BE:
case AF_FORMAT_FLOAT_LE:
ac->stream->codec->sample_fmt = AV_SAMPLE_FMT_FLT;
ac->sample_size = 4;
ac->sample_padding = sample_padding_float;
ao->format = AF_FORMAT_FLOAT_NE;
break;
}
// detect if we have to planarize
ac->planarize = false;
{
bool found_format = false;
bool found_planar_format = false;
for (sampleformat = codec->sample_fmts;
sampleformat && *sampleformat != AV_SAMPLE_FMT_NONE;
++sampleformat) {
if (*sampleformat == ac->stream->codec->sample_fmt)
found_format = true;
if (*sampleformat ==
av_get_planar_sample_fmt(ac->stream->codec->sample_fmt))
found_planar_format = true;
}
if (!found_format && found_planar_format) {
ac->stream->codec->sample_fmt =
av_get_planar_sample_fmt(ac->stream->codec->sample_fmt);
ac->planarize = true;
}
if (!found_format && !found_planar_format) {
// shouldn't happen
MP_ERR(ao, "sample format not found\n");
}
}
ac->stream->codec->bits_per_raw_sample = ac->sample_size * 8;
if (encode_lavc_open_codec(ao->encode_lavc_ctx, ac->stream) < 0)
return -1;
ac->pcmhack = 0;
if (ac->stream->codec->frame_size <= 1)
ac->pcmhack = av_get_bits_per_sample(ac->stream->codec->codec_id) / 8;
if (ac->pcmhack) {
ac->aframesize = 16384; // "enough"
ac->buffer_size =
ac->aframesize * ac->pcmhack * ao->channels.num * 2 + 200;
} else {
ac->aframesize = ac->stream->codec->frame_size;
ac->buffer_size =
ac->aframesize * ac->sample_size * ao->channels.num * 2 + 200;
}
if (ac->buffer_size < FF_MIN_BUFFER_SIZE)
ac->buffer_size = FF_MIN_BUFFER_SIZE;
ac->buffer = talloc_size(ac, ac->buffer_size);
// enough frames for at least 0.25 seconds
ac->framecount = ceil(ao->samplerate * 0.25 / ac->aframesize);
// but at least one!
ac->framecount = FFMAX(ac->framecount, 1);
ac->savepts = MP_NOPTS_VALUE;
ac->lastpts = MP_NOPTS_VALUE;
ac->offset = ac->stream->codec->sample_rate *
encode_lavc_getoffset(ao->encode_lavc_ctx, ac->stream);
ac->offset_left = ac->offset;
ao->untimed = true;
ao->priv = ac;
if (ac->planarize)
MP_WARN(ao, "need to planarize audio data\n");
return 0;
}
static void fill_with_padding(void *buf, int cnt, int sz, const void *padding)
{
int i;
if (sz == 1) {
memset(buf, cnt, *(char *)padding);
return;
}
for (i = 0; i < cnt; ++i)
memcpy((char *) buf + i * sz, padding, sz);
}
// close audio device
static void uninit(struct ao *ao, bool cut_audio)
{
struct encode_lavc_context *ectx = ao->encode_lavc_ctx;
if (!encode_lavc_start(ectx)) {
MP_WARN(ao, "not even ready to encode audio at end -> dropped");
return;
}
ao->priv = NULL;
}
// return: how many bytes can be played without blocking
static int get_space(struct ao *ao)
{
struct priv *ac = ao->priv;
return ac->aframesize * ac->framecount;
}
// must get exactly ac->aframesize amount of data
static int encode(struct ao *ao, double apts, void *data)
{
AVFrame *frame;
AVPacket packet;
struct priv *ac = ao->priv;
struct encode_lavc_context *ectx = ao->encode_lavc_ctx;
double realapts = ac->aframecount * (double) ac->aframesize /
ao->samplerate;
int status, gotpacket;
ac->aframecount++;
if (data)
ectx->audio_pts_offset = realapts - apts;
av_init_packet(&packet);
packet.data = ac->buffer;
packet.size = ac->buffer_size;
if(data)
{
frame = avcodec_alloc_frame();
frame->nb_samples = ac->aframesize;
if (ac->planarize) {
void *data2 = talloc_size(ao, ac->aframesize * ao->channels.num *
ac->sample_size);
reorder_to_planar(data2, data, ac->sample_size, ao->channels.num,
ac->aframesize);
data = data2;
}
size_t audiolen = ac->aframesize * ao->channels.num * ac->sample_size;
if (avcodec_fill_audio_frame(frame, ao->channels.num,
ac->stream->codec->sample_fmt, data,
audiolen, 1))
{
MP_ERR(ao, "error filling\n");
return -1;
}
if (ectx->options->rawts || ectx->options->copyts) {
// real audio pts
frame->pts = floor(apts * ac->stream->codec->time_base.den / ac->stream->codec->time_base.num + 0.5);
} else {
// audio playback time
frame->pts = floor(realapts * ac->stream->codec->time_base.den / ac->stream->codec->time_base.num + 0.5);
}
int64_t frame_pts = av_rescale_q(frame->pts, ac->stream->codec->time_base, ac->worst_time_base);
if (ac->lastpts != MP_NOPTS_VALUE && frame_pts <= ac->lastpts) {
// this indicates broken video
// (video pts failing to increase fast enough to match audio)
MP_WARN(ao, "audio frame pts went backwards (%d <- %d), autofixed\n",
(int)frame->pts, (int)ac->lastpts);
frame_pts = ac->lastpts + 1;
frame->pts = av_rescale_q(frame_pts, ac->worst_time_base, ac->stream->codec->time_base);
}
ac->lastpts = frame_pts;
frame->quality = ac->stream->codec->global_quality;
status = avcodec_encode_audio2(ac->stream->codec, &packet, frame, &gotpacket);
if (!status) {
if (ac->savepts == MP_NOPTS_VALUE)
ac->savepts = frame->pts;
}
avcodec_free_frame(&frame);
if (ac->planarize) {
talloc_free(data);
data = NULL;
}
}
else
{
status = avcodec_encode_audio2(ac->stream->codec, &packet, NULL, &gotpacket);
}
if(status) {
MP_ERR(ao, "error encoding\n");
return -1;
}
if(!gotpacket)
return 0;
MP_DBG(ao, "got pts %f (playback time: %f); out size: %d\n",
apts, realapts, packet.size);
encode_lavc_write_stats(ao->encode_lavc_ctx, ac->stream);
packet.stream_index = ac->stream->index;
// Do we need this at all? Better be safe than sorry...
if (packet.pts == AV_NOPTS_VALUE) {
MP_WARN(ao, "encoder lost pts, why?\n");
if (ac->savepts != MP_NOPTS_VALUE)
packet.pts = ac->savepts;
}
if (packet.pts != AV_NOPTS_VALUE)
packet.pts = av_rescale_q(packet.pts, ac->stream->codec->time_base,
ac->stream->time_base);
if (packet.dts != AV_NOPTS_VALUE)
packet.dts = av_rescale_q(packet.dts, ac->stream->codec->time_base,
ac->stream->time_base);
if(packet.duration > 0)
packet.duration = av_rescale_q(packet.duration, ac->stream->codec->time_base,
ac->stream->time_base);
ac->savepts = MP_NOPTS_VALUE;
if (encode_lavc_write_frame(ao->encode_lavc_ctx, &packet) < 0) {
MP_ERR(ao, "error writing at %f %f/%f\n",
realapts, (double) ac->stream->time_base.num,
(double) ac->stream->time_base.den);
return -1;
}
return packet.size;
}
// plays 'samples' samples of 'ni_data[0]'
// it should round it down to frame sizes
// return: number of samples played
static int play(struct ao *ao, void **ni_data, int samples, int flags)
{
struct priv *ac = ao->priv;
struct encode_lavc_context *ectx = ao->encode_lavc_ctx;
int bufpos = 0;
int64_t ptsoffset;
void *paddingbuf = NULL;
double nextpts;
double pts = ao->pts;
double outpts;
void *data = ni_data[0];
int len = samples * ao->sstride;
int bytelen = len;
len /= ac->sample_size * ao->channels.num;
if (!encode_lavc_start(ectx)) {
MP_WARN(ao, "not ready yet for encoding audio\n");
return 0;
}
if (flags & AOPLAY_FINAL_CHUNK) {
int written = 0;
if (len > 0) {
size_t extralen =
(ac->aframesize - 1) * ao->channels.num * ac->sample_size;
paddingbuf = talloc_size(NULL, bytelen + extralen);
memcpy(paddingbuf, data, bytelen);
fill_with_padding((char *) paddingbuf + bytelen,
extralen / ac->sample_size,
ac->sample_size, ac->sample_padding);
// No danger of recursion, because AOPLAY_FINAL_CHUNK not set
written =
play(ao, &paddingbuf, (bytelen + extralen) / ao->sstride, 0);
if (written * ao->sstride < bytelen) {
MP_ERR(ao, "did not write enough data at the end\n");
}
talloc_free(paddingbuf);
paddingbuf = NULL;
}
outpts = ac->expected_next_pts;
if (!ectx->options->rawts && ectx->options->copyts)
outpts += ectx->discontinuity_pts_offset;
outpts += encode_lavc_getoffset(ectx, ac->stream);
while (encode(ao, outpts, NULL) > 0) ;
return (FFMIN(written, bytelen)) / ao->sstride;
}
if (pts == MP_NOPTS_VALUE) {
MP_WARN(ao, "frame without pts, please report; synthesizing pts instead\n");
// synthesize pts from previous expected next pts
pts = ac->expected_next_pts;
}
if (ac->worst_time_base.den == 0) {
//if (ac->stream->codec->time_base.num / ac->stream->codec->time_base.den >= ac->stream->time_base.num / ac->stream->time_base.den)
if (ac->stream->codec->time_base.num * (double) ac->stream->time_base.den >=
ac->stream->time_base.num * (double) ac->stream->codec->time_base.den) {
MP_VERBOSE(ao, "NOTE: using codec time base (%d/%d) for pts "
"adjustment; the stream base (%d/%d) is not worse.\n",
(int)ac->stream->codec->time_base.num,
(int)ac->stream->codec->time_base.den,
(int)ac->stream->time_base.num,
(int)ac->stream->time_base.den);
ac->worst_time_base = ac->stream->codec->time_base;
ac->worst_time_base_is_stream = 0;
} else {
MP_WARN(ao, "NOTE: not using codec time base (%d/%d) for pts "
"adjustment; the stream base (%d/%d) is worse.\n",
(int)ac->stream->codec->time_base.num,
(int)ac->stream->codec->time_base.den,
(int)ac->stream->time_base.num,
(int)ac->stream->time_base.den);
ac->worst_time_base = ac->stream->time_base;
ac->worst_time_base_is_stream = 1;
}
// NOTE: we use the following "axiom" of av_rescale_q:
// if time base A is worse than time base B, then
// av_rescale_q(av_rescale_q(x, A, B), B, A) == x
// this can be proven as long as av_rescale_q rounds to nearest, which
// it currently does
// av_rescale_q(x, A, B) * B = "round x*A to nearest multiple of B"
// and:
// av_rescale_q(av_rescale_q(x, A, B), B, A) * A
// == "round av_rescale_q(x, A, B)*B to nearest multiple of A"
// == "round 'round x*A to nearest multiple of B' to nearest multiple of A"
//
// assume this fails. Then there is a value of x*A, for which the
// nearest multiple of B is outside the range [(x-0.5)*A, (x+0.5)*A[.
// Absurd, as this range MUST contain at least one multiple of B.
}
ptsoffset = ac->offset;
// this basically just edits ao->apts for syncing purposes
if (ectx->options->copyts || ectx->options->rawts) {
// we do not send time sync data to the video side,
// but we always need the exact pts, even if zero
} else {
// here we must "simulate" the pts editing
// 1. if we have to skip stuff, we skip it
// 2. if we have to add samples, we add them
// 3. we must still adjust ptsoffset appropriately for AV sync!
// invariant:
// if no partial skipping is done, the first frame gets ao->apts passed as pts!
if (ac->offset_left < 0) {
if (ac->offset_left <= -len) {
// skip whole frame
ac->offset_left += len;
return len;
} else {
// skip part of this frame, buffer/encode the rest
bufpos -= ac->offset_left;
ptsoffset += ac->offset_left;
ac->offset_left = 0;
}
} else if (ac->offset_left > 0) {
// make a temporary buffer, filled with zeroes at the start
// (don't worry, only happens once)
paddingbuf = talloc_size(ac, ac->sample_size * ao->channels.num *
(ac->offset_left + len));
fill_with_padding(paddingbuf, ac->offset_left, ac->sample_size,
ac->sample_padding);
data = (char *) paddingbuf + ac->sample_size * ao->channels.num *
ac->offset_left;
bufpos -= ac->offset_left; // yes, negative!
ptsoffset += ac->offset_left;
ac->offset_left = 0;
// now adjust the bufpos so the final value of bufpos is positive!
/*
int cnt = (len - bufpos) / ac->aframesize;
int finalbufpos = bufpos + cnt * ac->aframesize;
*/
int finalbufpos = len - (len - bufpos) % ac->aframesize;
if (finalbufpos < 0) {
MP_WARN(ao, "cannot attain the "
"exact requested audio sync; shifting by %d frames\n",
-finalbufpos);
bufpos -= finalbufpos;
}
}
}
if (!ectx->options->rawts && ectx->options->copyts) {
// fix the discontinuity pts offset
nextpts = pts + ptsoffset / (double) ao->samplerate;
if (ectx->discontinuity_pts_offset == MP_NOPTS_VALUE) {
ectx->discontinuity_pts_offset = ectx->next_in_pts - nextpts;
}
else if (fabs(nextpts + ectx->discontinuity_pts_offset - ectx->next_in_pts) > 30) {
MP_WARN(ao, "detected an unexpected discontinuity (pts jumped by "
"%f seconds)\n",
nextpts + ectx->discontinuity_pts_offset - ectx->next_in_pts);
ectx->discontinuity_pts_offset = ectx->next_in_pts - nextpts;
}
outpts = pts + ectx->discontinuity_pts_offset;
}
else
outpts = pts;
while (len - bufpos >= ac->aframesize) {
encode(ao,
outpts + (bufpos + ptsoffset) / (double) ao->samplerate + encode_lavc_getoffset(ectx, ac->stream),
(char *) data + ac->sample_size * bufpos * ao->channels.num);
bufpos += ac->aframesize;
}
talloc_free(paddingbuf);
// calculate expected pts of next audio frame
ac->expected_next_pts = pts + (bufpos + ptsoffset) / (double) ao->samplerate;
if (!ectx->options->rawts && ectx->options->copyts) {
// set next allowed output pts value
nextpts = ac->expected_next_pts + ectx->discontinuity_pts_offset;
if (nextpts > ectx->next_in_pts)
ectx->next_in_pts = nextpts;
}
return bufpos;
}
const struct ao_driver audio_out_lavc = {
.encode = true,
.description = "audio encoding using libavcodec",
.name = "lavc",
.init = init,
.uninit = uninit,
.get_space = get_space,
.play = play,
};
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