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mpv/audio/out/ao_lavc.c
wm4 68ff8a0484 Move compat/ and bstr/ directory contents somewhere else 
bstr.c doesn't really deserve its own directory, and compat had just
a few files, most of which may as well be in osdep. There isn't really
any justification for these extra directories, so get rid of them.

The compat/libav.h was empty - just delete it. We changed our approach
to API compatibility, and will likely not need it anymore.
2014-08-29 12:31:52 +02:00

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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 <assert.h>
#include <limits.h>
#include <libavutil/common.h>
#include <libavutil/audioconvert.h>
#include "config.h"
#include "options/options.h"
#include "common/common.h"
#include "audio/format.h"
#include "audio/fmt-conversion.h"
#include "talloc.h"
#include "ao.h"
#include "internal.h"
#include "common/msg.h"
#include "common/encode_lavc.h"
struct priv {
uint8_t *buffer;
size_t buffer_size;
AVStream *stream;
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;
};
static void select_format(struct ao *ao, AVCodec *codec)
{
int best_score = INT_MIN;
int best_format = 0;
// Check the encoder's list of supported formats.
for (const enum AVSampleFormat *sampleformat = codec->sample_fmts;
sampleformat && *sampleformat != AV_SAMPLE_FMT_NONE;
++sampleformat)
{
int fmt = af_from_avformat(*sampleformat);
if (!fmt) {
MP_WARN(ao, "unsupported lavc format %s\n",
av_get_sample_fmt_name(*sampleformat));
continue;
}
int score = af_format_conversion_score(fmt, ao->format);
if (score > best_score) {
best_score = score;
best_format = fmt;
}
}
if (best_format) {
ao->format = best_format;
} else {
MP_ERR(ao, "sample format not found\n"); // shouldn't happen
}
}
// 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);
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");
goto fail;
}
codec = encode_lavc_get_codec(ao->encode_lavc_ctx, ac->stream);
// 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->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))
goto fail;
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;
select_format(ao, codec);
ac->sample_size = af_fmt2bps(ao->format);
ac->stream->codec->sample_fmt = af_to_avformat(ao->format);
ac->stream->codec->bits_per_raw_sample = ac->sample_size * 8;
if (encode_lavc_open_codec(ao->encode_lavc_ctx, ac->stream) < 0)
goto fail;
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 = AV_NOPTS_VALUE;
ac->lastpts = AV_NOPTS_VALUE;
ao->untimed = true;
pthread_mutex_unlock(&ao->encode_lavc_ctx->lock);
return 0;
fail:
pthread_mutex_unlock(&ao->encode_lavc_ctx->lock);
return -1;
}
// close audio device
static int 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;
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->buffer) {
double 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) ;
}
pthread_mutex_unlock(&ectx->lock);
}
// 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)
{
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) {
AVFrame *frame = av_frame_alloc();
frame->format = af_to_avformat(ao->format);
frame->nb_samples = ac->aframesize;
assert(ao->channels.num <= AV_NUM_DATA_POINTERS);
for (int n = 0; n < ao->channels.num; 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->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 != 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->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 == AV_NOPTS_VALUE)
ac->savepts = frame->pts;
}
av_frame_free(&frame);
}
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 = AV_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;
}
// 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;
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;
void *padded[MP_NUM_CHANNELS];
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->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.
}
// 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->stream);
while (samples - bufpos >= ac->aframesize) {
void *start[MP_NUM_CHANNELS];
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,
};
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