RepoMirrors/mpv
1
0
Fork 0
mirror of https://github.com/mpv-player/mpv synced 2024-12-19 13:21:13 +00:00
Code Issues Releases Wiki Activity
b016760a28
mpv/audio/out/ao_lavc.c
wm4 037c37519b audio/out: require AO drivers to report period size and correct buffer 
Before this change, AOs could have internal alignment, and play() would
not consume the trailing data if the size passed to it is not aligned.
Change this to require AOs to report their alignment (via period_size),
and make sure to always send aligned data.

The buffer reported by get_space() now always has to be correct and
reliable. If play() does not consume all data provided (which is bounded
by get_space()), an error is printed.

This is preparation for potential further AO changes.

I casually checked alsa/lavc/null/pcm, the other AOs might or might not
work.
2017-06-25 15:57:43 +02:00

520 lines
17 KiB
C
Raw Blame History

/*
* 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];
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
Reference in New Issue View Git Blame Copy Permalink