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028e51d8af
mpv/audio/out/ao_lavc.c
wm4 d36ff64b29 audio: fix annyoing af_get_best_sample_formats() definition 
The af_get_best_sample_formats() function had an argument of
int[AF_FORMAT_COUNT], which is slightly incorrect, because it's 0
terminated and should in theory have AF_FORMAT_COUNT+1 entries. It won't
actually write this many formats (since some formats are fundamentally
incompatible), but it still feels annoying and incorrect. So fix it, and
require that callers pass an AF_FORMAT_COUNT+1 array.

Note that the array size has no meaning in C function arguments (just
another issue with C static arrays being weird and stupid), so get rid
of it completely.

Not changing the af_lavcac3enc use, since that is rewritten in another
branch anyway.
2018-01-25 20:18:32 -08:00

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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.
*
* mpv is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* mpv 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 Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with mpv. If not, see <http://www.gnu.org/licenses/>.
*/
#include <stdio.h>
#include <stdlib.h>
#include <assert.h>
#include <limits.h>
#include <libavutil/common.h>
#include "config.h"
#include "options/options.h"
#include "common/common.h"
#include "audio/format.h"
#include "audio/fmt-conversion.h"
#include "mpv_talloc.h"
#include "ao.h"
#include "internal.h"
#include "common/msg.h"
#include "common/encode_lavc.h"
struct priv {
AVStream *stream;
AVCodecContext *codec;
int pcmhack;
int aframesize;
int aframecount;
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;
bool shutdown;
};
static bool supports_format(AVCodec *codec, int format)
{
for (const enum AVSampleFormat *sampleformat = codec->sample_fmts;
sampleformat && *sampleformat != AV_SAMPLE_FMT_NONE;
++sampleformat)
{
if (af_from_avformat(*sampleformat) == format)
return true;
}
return false;
}
static void select_format(struct ao *ao, AVCodec *codec)
{
int formats[AF_FORMAT_COUNT + 1];
af_get_best_sample_formats(ao->format, formats);
for (int n = 0; formats[n]; n++) {
if (supports_format(codec, formats[n])) {
ao->format = formats[n];
break;
}
}
}
// open & setup audio device
static int init(struct ao *ao)
{
struct priv *ac = talloc_zero(ao, struct priv);
AVCodec *codec;
ao->priv = ac;
if (!encode_lavc_available(ao->encode_lavc_ctx)) {
MP_ERR(ao, "the option --o (output file) must be specified\n");
return -1;
}
pthread_mutex_lock(&ao->encode_lavc_ctx->lock);
if (encode_lavc_alloc_stream(ao->encode_lavc_ctx,
AVMEDIA_TYPE_AUDIO,
&ac->stream, &ac->codec) < 0) {
MP_ERR(ao, "could not get a new audio stream\n");
goto fail;
}
codec = ao->encode_lavc_ctx->ac;
int samplerate = af_select_best_samplerate(ao->samplerate,
codec->supported_samplerates);
if (samplerate > 0)
ao->samplerate = samplerate;
// TODO: Remove this redundancy with encode_lavc_alloc_stream also
// setting the time base.
// Using codec->time_bvase is deprecated, but needed for older lavf.
ac->stream->time_base.num = 1;
ac->stream->time_base.den = ao->samplerate;
ac->codec->time_base.num = 1;
ac->codec->time_base.den = ao->samplerate;
ac->codec->sample_rate = ao->samplerate;
struct mp_chmap_sel sel = {0};
mp_chmap_sel_add_any(&sel);
if (!ao_chmap_sel_adjust2(ao, &sel, &ao->channels, false))
goto fail;
mp_chmap_reorder_to_lavc(&ao->channels);
ac->codec->channels = ao->channels.num;
ac->codec->channel_layout = mp_chmap_to_lavc(&ao->channels);
ac->codec->sample_fmt = AV_SAMPLE_FMT_NONE;
select_format(ao, codec);
ac->sample_size = af_fmt_to_bytes(ao->format);
ac->codec->sample_fmt = af_to_avformat(ao->format);
ac->codec->bits_per_raw_sample = ac->sample_size * 8;
if (encode_lavc_open_codec(ao->encode_lavc_ctx, ac->codec) < 0)
goto fail;
ac->pcmhack = 0;
if (ac->codec->frame_size <= 1)
ac->pcmhack = av_get_bits_per_sample(ac->codec->codec_id) / 8;
if (ac->pcmhack)
ac->aframesize = 16384; // "enough"
else
ac->aframesize = ac->codec->frame_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 = AV_NOPTS_VALUE;
ac->lastpts = AV_NOPTS_VALUE;
ao->untimed = true;
ao->period_size = ac->aframesize * ac->framecount;
if (ao->channels.num > AV_NUM_DATA_POINTERS)
goto fail;
pthread_mutex_unlock(&ao->encode_lavc_ctx->lock);
return 0;
fail:
pthread_mutex_unlock(&ao->encode_lavc_ctx->lock);
ac->shutdown = true;
return -1;
}
// close audio device
static void encode(struct ao *ao, double apts, void **data);
static void uninit(struct ao *ao)
{
struct priv *ac = ao->priv;
struct encode_lavc_context *ectx = ao->encode_lavc_ctx;
if (!ac || ac->shutdown)
return;
pthread_mutex_lock(&ectx->lock);
if (!encode_lavc_start(ectx)) {
MP_WARN(ao, "not even ready to encode audio at end -> dropped\n");
pthread_mutex_unlock(&ectx->lock);
return;
}
if (ac->stream) {
double outpts = ac->expected_next_pts;
if (!ectx->options->rawts && ectx->options->copyts)
outpts += ectx->discontinuity_pts_offset;
outpts += encode_lavc_getoffset(ectx, ac->codec);
encode(ao, outpts, NULL);
}
pthread_mutex_unlock(&ectx->lock);
ac->shutdown = true;
}
// return: how many samples can be played without blocking
static int get_space(struct ao *ao)
{
struct priv *ac = ao->priv;
return ac->aframesize * ac->framecount;
}
static void write_packet(struct ao *ao, AVPacket *packet)
{
// TODO: Can we unify this with the equivalent video code path?
struct priv *ac = ao->priv;
packet->stream_index = ac->stream->index;
if (packet->pts != AV_NOPTS_VALUE) {
packet->pts = av_rescale_q(packet->pts,
ac->codec->time_base,
ac->stream->time_base);
} else {
// Do we need this at all? Better be safe than sorry...
MP_WARN(ao, "encoder lost pts, why?\n");
if (ac->savepts != MP_NOPTS_VALUE) {
packet->pts = av_rescale_q(ac->savepts,
ac->codec->time_base,
ac->stream->time_base);
}
}
if (packet->dts != AV_NOPTS_VALUE) {
packet->dts = av_rescale_q(packet->dts,
ac->codec->time_base,
ac->stream->time_base);
}
if (packet->duration > 0) {
packet->duration = av_rescale_q(packet->duration,
ac->codec->time_base,
ac->stream->time_base);
}
ac->savepts = AV_NOPTS_VALUE;
if (encode_lavc_write_frame(ao->encode_lavc_ctx,
ac->stream, packet) < 0) {
MP_ERR(ao, "error writing at %d %d/%d\n",
(int) packet->pts,
ac->stream->time_base.num,
ac->stream->time_base.den);
return;
}
}
static void encode_audio_and_write(struct ao *ao, AVFrame *frame)
{
// TODO: Can we unify this with the equivalent video code path?
struct priv *ac = ao->priv;
AVPacket packet = {0};
int status = avcodec_send_frame(ac->codec, frame);
if (status < 0) {
MP_ERR(ao, "error encoding at %d %d/%d\n",
frame ? (int) frame->pts : -1,
ac->codec->time_base.num,
ac->codec->time_base.den);
return;
}
for (;;) {
av_init_packet(&packet);
status = avcodec_receive_packet(ac->codec, &packet);
if (status == AVERROR(EAGAIN)) { // No more packets for now.
if (frame == NULL) {
MP_ERR(ao, "sent flush frame, got EAGAIN");
}
break;
}
if (status == AVERROR_EOF) { // No more packets, ever.
if (frame != NULL) {
MP_ERR(ao, "sent audio frame, got EOF");
}
break;
}
if (status < 0) {
MP_ERR(ao, "error encoding at %d %d/%d\n",
frame ? (int) frame->pts : -1,
ac->codec->time_base.num,
ac->codec->time_base.den);
break;
}
if (frame) {
if (ac->savepts == AV_NOPTS_VALUE)
ac->savepts = frame->pts;
}
encode_lavc_write_stats(ao->encode_lavc_ctx, ac->codec);
write_packet(ao, &packet);
av_packet_unref(&packet);
}
}
// must get exactly ac->aframesize amount of data
static void encode(struct ao *ao, double apts, void **data)
{
struct priv *ac = ao->priv;
struct encode_lavc_context *ectx = ao->encode_lavc_ctx;
double realapts = ac->aframecount * (double) ac->aframesize /
ao->samplerate;
ac->aframecount++;
if (data)
ectx->audio_pts_offset = realapts - apts;
if(data) {
AVFrame *frame = av_frame_alloc();
frame->format = af_to_avformat(ao->format);
frame->nb_samples = ac->aframesize;
size_t num_planes = af_fmt_is_planar(ao->format) ? ao->channels.num : 1;
assert(num_planes <= AV_NUM_DATA_POINTERS);
for (int n = 0; n < num_planes; n++)
frame->extended_data[n] = data[n];
frame->linesize[0] = frame->nb_samples * ao->sstride;
if (ectx->options->rawts || ectx->options->copyts) {
// real audio pts
frame->pts = floor(apts * ac->codec->time_base.den / ac->codec->time_base.num + 0.5);
} else {
// audio playback time
frame->pts = floor(realapts * ac->codec->time_base.den / ac->codec->time_base.num + 0.5);
}
int64_t frame_pts = av_rescale_q(frame->pts, ac->codec->time_base, ac->worst_time_base);
if (ac->lastpts != AV_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->codec->time_base);
}
ac->lastpts = frame_pts;
frame->quality = ac->codec->global_quality;
encode_audio_and_write(ao, frame);
av_frame_free(&frame);
}
else
encode_audio_and_write(ao, NULL);
}
// this should round samples down to frame sizes
// return: number of samples played
static int play(struct ao *ao, void **data, int samples, int flags)
{
struct priv *ac = ao->priv;
struct encode_lavc_context *ectx = ao->encode_lavc_ctx;
int bufpos = 0;
double nextpts;
double outpts;
int orig_samples = samples;
pthread_mutex_lock(&ectx->lock);
if (!encode_lavc_start(ectx)) {
MP_WARN(ao, "not ready yet for encoding audio\n");
pthread_mutex_unlock(&ectx->lock);
return 0;
}
double pts = ectx->last_audio_in_pts;
pts += ectx->samples_since_last_pts / (double)ao->samplerate;
size_t num_planes = af_fmt_is_planar(ao->format) ? ao->channels.num : 1;
void *tempdata = NULL;
void *padded[MP_NUM_CHANNELS];
if ((flags & AOPLAY_FINAL_CHUNK) && (samples % ac->aframesize)) {
tempdata = talloc_new(NULL);
size_t bytelen = samples * ao->sstride;
size_t extralen = (ac->aframesize - 1) * ao->sstride;
for (int n = 0; n < num_planes; n++) {
padded[n] = talloc_size(tempdata, bytelen + extralen);
memcpy(padded[n], data[n], bytelen);
af_fill_silence((char *)padded[n] + bytelen, extralen, ao->format);
}
data = padded;
samples = (bytelen + extralen) / 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->codec->time_base.num / ac->codec->time_base.den >= ac->stream->time_base.num / ac->stream->time_base.den)
if (ac->codec->time_base.num * (double) ac->stream->time_base.den >=
ac->stream->time_base.num * (double) ac->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->codec->time_base.num,
(int)ac->codec->time_base.den,
(int)ac->stream->time_base.num,
(int)ac->stream->time_base.den);
ac->worst_time_base = ac->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->codec->time_base.num,
(int)ac->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.
}
// Fix and apply the discontinuity pts offset.
if (!ectx->options->rawts && ectx->options->copyts) {
// fix the discontinuity pts offset
nextpts = pts;
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;
}
// Shift pts by the pts offset first.
outpts += encode_lavc_getoffset(ectx, ac->codec);
while (samples - bufpos >= ac->aframesize) {
void *start[MP_NUM_CHANNELS] = {0};
for (int n = 0; n < num_planes; n++)
start[n] = (char *)data[n] + bufpos * ao->sstride;
encode(ao, outpts + bufpos / (double) ao->samplerate, start);
bufpos += ac->aframesize;
}
// Calculate expected pts of next audio frame (input side).
ac->expected_next_pts = pts + bufpos / (double) ao->samplerate;
// Set next allowed input pts value (input side).
if (!ectx->options->rawts && ectx->options->copyts) {
nextpts = ac->expected_next_pts + ectx->discontinuity_pts_offset;
if (nextpts > ectx->next_in_pts)
ectx->next_in_pts = nextpts;
}
talloc_free(tempdata);
int taken = FFMIN(bufpos, orig_samples);
ectx->samples_since_last_pts += taken;
pthread_mutex_unlock(&ectx->lock);
if (flags & AOPLAY_FINAL_CHUNK) {
if (bufpos < orig_samples) {
MP_ERR(ao, "did not write enough data at the end\n");
}
} else {
if (bufpos > orig_samples) {
MP_ERR(ao, "audio buffer overflow (should never happen)\n");
}
}
return taken;
}
static void drain(struct ao *ao)
{
// pretend we support it, so generic code doesn't force a wait
}
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,
.drain = drain,
};
// vim: sw=4 ts=4 et tw=80
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