mpv/audio/decode/ad_lavc.c

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/*
* This file is part of mpv.
*
dec_audio, ad_lavc: change license to LGPL All relevant authors of the current code have agreed. As always, there are the usual historical artifacts that could be mentioned. For example, there used to be a large number of decoders by various authors who were not asked, but whose code was all 100% removed. (Mostly due to FFmpeg providing all codecs.) One point of contention is that Nick Kurshev might have refactored the old audio decoder code in 2001. Basically, there are hints that it might have been done by him, such as Arpi's commit message stating that the code was imported from MPlayerXP (Nick's fork), or all the files having his name in the "maintainer" field. On the other hand, the murky history of ad.h weakens this - it could be that Arpi started this work, and Nick took it (and possibly finished it). In any case, Nick could not be reached, so there is no agreement for LGPL relicensing from him. We're changing the license anyway, and assume that his change in itself is not copyrightable. He only moved code, and in addition used the equivalent video decoder framework (done by Arpi, who agreed) as template. For example, ad_functions_s was basically vd_functions_s, which the signature of the decode callback changed to the same as audio_decode(). ad_functions_s also had a comment that said it interfaces with "video decoder drivers" (I'm fixing this comment in this commit). I verified that no additional code was added that is copyright-relevant, still in today's code, and not copied from the existing code at the time (either from the previous audio decoder code or the video framework code). What apparently matters here is that none of the old code was not written by Nick, and the authors of the old code have given his agreement, and (probably) that Nick didn't add actual new code (none that would have survived), that was not trivially based on the old one (i.e. no new copyrightable "work"). A copyright expert told me that this kind of change can be considered not relevant for copyright, so here we go. Rewriting this would end with the same code anyway, and the naming conventions can't be copyrighted.
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* 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
dec_audio, ad_lavc: change license to LGPL All relevant authors of the current code have agreed. As always, there are the usual historical artifacts that could be mentioned. For example, there used to be a large number of decoders by various authors who were not asked, but whose code was all 100% removed. (Mostly due to FFmpeg providing all codecs.) One point of contention is that Nick Kurshev might have refactored the old audio decoder code in 2001. Basically, there are hints that it might have been done by him, such as Arpi's commit message stating that the code was imported from MPlayerXP (Nick's fork), or all the files having his name in the "maintainer" field. On the other hand, the murky history of ad.h weakens this - it could be that Arpi started this work, and Nick took it (and possibly finished it). In any case, Nick could not be reached, so there is no agreement for LGPL relicensing from him. We're changing the license anyway, and assume that his change in itself is not copyrightable. He only moved code, and in addition used the equivalent video decoder framework (done by Arpi, who agreed) as template. For example, ad_functions_s was basically vd_functions_s, which the signature of the decode callback changed to the same as audio_decode(). ad_functions_s also had a comment that said it interfaces with "video decoder drivers" (I'm fixing this comment in this commit). I verified that no additional code was added that is copyright-relevant, still in today's code, and not copied from the existing code at the time (either from the previous audio decoder code or the video framework code). What apparently matters here is that none of the old code was not written by Nick, and the authors of the old code have given his agreement, and (probably) that Nick didn't add actual new code (none that would have survived), that was not trivially based on the old one (i.e. no new copyrightable "work"). A copyright expert told me that this kind of change can be considered not relevant for copyright, so here we go. Rewriting this would end with the same code anyway, and the naming conventions can't be copyrighted.
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* GNU Lesser General Public License for more details.
*
dec_audio, ad_lavc: change license to LGPL All relevant authors of the current code have agreed. As always, there are the usual historical artifacts that could be mentioned. For example, there used to be a large number of decoders by various authors who were not asked, but whose code was all 100% removed. (Mostly due to FFmpeg providing all codecs.) One point of contention is that Nick Kurshev might have refactored the old audio decoder code in 2001. Basically, there are hints that it might have been done by him, such as Arpi's commit message stating that the code was imported from MPlayerXP (Nick's fork), or all the files having his name in the "maintainer" field. On the other hand, the murky history of ad.h weakens this - it could be that Arpi started this work, and Nick took it (and possibly finished it). In any case, Nick could not be reached, so there is no agreement for LGPL relicensing from him. We're changing the license anyway, and assume that his change in itself is not copyrightable. He only moved code, and in addition used the equivalent video decoder framework (done by Arpi, who agreed) as template. For example, ad_functions_s was basically vd_functions_s, which the signature of the decode callback changed to the same as audio_decode(). ad_functions_s also had a comment that said it interfaces with "video decoder drivers" (I'm fixing this comment in this commit). I verified that no additional code was added that is copyright-relevant, still in today's code, and not copied from the existing code at the time (either from the previous audio decoder code or the video framework code). What apparently matters here is that none of the old code was not written by Nick, and the authors of the old code have given his agreement, and (probably) that Nick didn't add actual new code (none that would have survived), that was not trivially based on the old one (i.e. no new copyrightable "work"). A copyright expert told me that this kind of change can be considered not relevant for copyright, so here we go. Rewriting this would end with the same code anyway, and the naming conventions can't be copyrighted.
2017-06-14 14:20:56 +00:00
* 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 <unistd.h>
#include <stdbool.h>
#include <assert.h>
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#include <libavcodec/avcodec.h>
#include <libavutil/opt.h>
#include <libavutil/common.h>
#include <libavutil/intreadwrite.h>
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#include "mpv_talloc.h"
#include "audio/aframe.h"
#include "audio/fmt-conversion.h"
#include "common/av_common.h"
#include "common/codecs.h"
#include "common/global.h"
#include "common/msg.h"
#include "demux/packet.h"
#include "demux/stheader.h"
#include "filters/f_decoder_wrapper.h"
#include "filters/filter_internal.h"
#include "options/m_config.h"
#include "options/options.h"
struct priv {
AVCodecContext *avctx;
AVFrame *avframe;
struct mp_chmap force_channel_map;
uint32_t skip_samples, trim_samples;
bool preroll_done;
double next_pts;
AVRational codec_timebase;
ad_lavc, vd_lavc: return full error codes to shared decoder loop ad_lavc and vd_lavc use the lavc_process() helper to translate the FFmpeg push/pull API to the internal filter API (which completely mismatch, even though I'm responsible for both, just fucking kill me). This interface was "slightly" too tight. It returned only a bool indicating "progress", which was not enough to handle some cases (see following commit). While we're at it, move all state into a struct. This is only a single bool, but we get the chance to add more if needed. This fixes mpv falling asleep if decoding returns an error during draining. If decoding fails when we already sent EOF, the state machine stopped making progress. This left mpv just sitting around and doing nothing. A test case can be created with: echo $RANDOM >> image.png This makes libavformat read a proper packet plus a packet of garbage. libavcodec will decode a frame, and then return an error code. The lavc_process() wrapper could not deal with this, because there was no way to differentiate between "retry" and "send new packet". Normally, it would send a new packet, so decoding would make progress anyway. If there was "progress", we couldn't just retry, because it'd retry forever. This is made worse by the fact that it tries to decode at least two frames before starting display, meaning it will "sit around and do nothing" before the picture is displayed. Change it so that on error return, "receiving" a frame is retried. This will make it return the EOF, so everything works properly. This is a high-risk change, because all these funny bullshit exceptions for hardware decoding are in the way, and I didn't retest them. For example, if hardware decoding is enabled, it keeps a list of packets, that are fed into the decoder again if hardware decoding fails, and a software fallback is performed. Another case of horrifying accidental complexity. Fixes: #6618
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struct lavc_state state;
struct mp_decoder public;
};
#define OPT_BASE_STRUCT struct ad_lavc_params
struct ad_lavc_params {
float ac3drc;
int downmix;
int threads;
char **avopts;
};
const struct m_sub_options ad_lavc_conf = {
.opts = (const m_option_t[]) {
{"ac3drc", OPT_FLOAT(ac3drc), M_RANGE(0, 6)},
{"downmix", OPT_FLAG(downmix)},
{"threads", OPT_INT(threads), M_RANGE(0, 16)},
{"o", OPT_KEYVALUELIST(avopts)},
{0}
},
.size = sizeof(struct ad_lavc_params),
.defaults = &(const struct ad_lavc_params){
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.ac3drc = 0,
.downmix = 0,
.threads = 1,
},
};
static bool init(struct mp_filter *da, struct mp_codec_params *codec,
const char *decoder)
{
struct priv *ctx = da->priv;
struct MPOpts *mpopts = mp_get_config_group(ctx, da->global, &mp_opt_root);
struct ad_lavc_params *opts =
mp_get_config_group(ctx, da->global, &ad_lavc_conf);
AVCodecContext *lavc_context;
const AVCodec *lavc_codec;
ctx->codec_timebase = mp_get_codec_timebase(codec);
if (codec->force_channels)
ctx->force_channel_map = codec->channels;
core: redo how codecs are mapped, remove codecs.conf Use codec names instead of FourCCs to identify codecs. Rewrite how codecs are selected and initialized. Now each decoder exports a list of decoders (and the codec it supports) via add_decoders(). The order matters, and the first decoder for a given decoder is preferred over the other decoders. E.g. all ad_mpg123 decoders are preferred over ad_lavc, because it comes first in the mpcodecs_ad_drivers array. Likewise, decoders within ad_lavc that are enumerated first by libavcodec (using av_codec_next()) are preferred. (This is actually critical to select h264 software decoding by default instead of vdpau. libavcodec and ffmpeg/avconv use the same method to select decoders by default, so we hope this is sane.) The codec names follow libavcodec's codec names as defined by AVCodecDescriptor.name (see libavcodec/codec_desc.c). Some decoders have names different from the canonical codec name. The AVCodecDescriptor API is relatively new, so we need a compatibility layer for older libavcodec versions for codec names that are referenced internally, and which are different from the decoder name. (Add a configure check for that, because checking versions is getting way too messy.) demux/codec_tags.c is generated from the former codecs.conf (minus "special" decoders like vdpau, and excluding the mappings that are the same as the mappings libavformat's exported RIFF tables). It contains all the mappings from FourCCs to codec name. This is needed for demux_mkv, demux_mpg, demux_avi and demux_asf. demux_lavf will set the codec as determined by libavformat, while the other demuxers have to do this on their own, using the mp_set_audio/video_codec_from_tag() functions. Note that the sh_audio/video->format members don't uniquely identify the codec anymore, and sh->codec takes over this role. Replace the --ac/--vc/--afm/--vfm with new --vd/--ad options, which provide cover the functionality of the removed switched. Note: there's no CODECS_FLAG_FLIP flag anymore. This means some obscure container/video combinations (e.g. the sample Film_200_zygo_pro.mov) are played flipped. ffplay/avplay doesn't handle this properly either, so we don't care and blame ffmeg/libav instead.
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lavc_codec = avcodec_find_decoder_by_name(decoder);
if (!lavc_codec) {
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MP_ERR(da, "Cannot find codec '%s' in libavcodec...\n", decoder);
return false;
}
lavc_context = avcodec_alloc_context3(lavc_codec);
ctx->avctx = lavc_context;
ctx->avframe = av_frame_alloc();
lavc_context->codec_type = AVMEDIA_TYPE_AUDIO;
lavc_context->codec_id = lavc_codec->id;
lavc_context->pkt_timebase = ctx->codec_timebase;
if (opts->downmix && mpopts->audio_output_channels.num_chmaps == 1) {
lavc_context->request_channel_layout =
mp_chmap_to_lavc(&mpopts->audio_output_channels.chmaps[0]);
}
// Always try to set - option only exists for AC3 at the moment
av_opt_set_double(lavc_context, "drc_scale", opts->ac3drc,
AV_OPT_SEARCH_CHILDREN);
// Let decoder add AV_FRAME_DATA_SKIP_SAMPLES.
av_opt_set(lavc_context, "flags2", "+skip_manual", AV_OPT_SEARCH_CHILDREN);
mp_set_avopts(da->log, lavc_context, opts->avopts);
if (mp_set_avctx_codec_headers(lavc_context, codec) < 0) {
MP_ERR(da, "Could not set decoder parameters.\n");
return false;
}
mp_set_avcodec_threads(da->log, lavc_context, opts->threads);
/* open it */
if (avcodec_open2(lavc_context, lavc_codec, NULL) < 0) {
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MP_ERR(da, "Could not open codec.\n");
return false;
}
ctx->next_pts = MP_NOPTS_VALUE;
return true;
}
static void destroy(struct mp_filter *da)
{
struct priv *ctx = da->priv;
avcodec_free_context(&ctx->avctx);
av_frame_free(&ctx->avframe);
}
static void reset(struct mp_filter *da)
{
struct priv *ctx = da->priv;
avcodec_flush_buffers(ctx->avctx);
ctx->skip_samples = 0;
ctx->trim_samples = 0;
ctx->preroll_done = false;
ctx->next_pts = MP_NOPTS_VALUE;
ad_lavc, vd_lavc: return full error codes to shared decoder loop ad_lavc and vd_lavc use the lavc_process() helper to translate the FFmpeg push/pull API to the internal filter API (which completely mismatch, even though I'm responsible for both, just fucking kill me). This interface was "slightly" too tight. It returned only a bool indicating "progress", which was not enough to handle some cases (see following commit). While we're at it, move all state into a struct. This is only a single bool, but we get the chance to add more if needed. This fixes mpv falling asleep if decoding returns an error during draining. If decoding fails when we already sent EOF, the state machine stopped making progress. This left mpv just sitting around and doing nothing. A test case can be created with: echo $RANDOM >> image.png This makes libavformat read a proper packet plus a packet of garbage. libavcodec will decode a frame, and then return an error code. The lavc_process() wrapper could not deal with this, because there was no way to differentiate between "retry" and "send new packet". Normally, it would send a new packet, so decoding would make progress anyway. If there was "progress", we couldn't just retry, because it'd retry forever. This is made worse by the fact that it tries to decode at least two frames before starting display, meaning it will "sit around and do nothing" before the picture is displayed. Change it so that on error return, "receiving" a frame is retried. This will make it return the EOF, so everything works properly. This is a high-risk change, because all these funny bullshit exceptions for hardware decoding are in the way, and I didn't retest them. For example, if hardware decoding is enabled, it keeps a list of packets, that are fed into the decoder again if hardware decoding fails, and a software fallback is performed. Another case of horrifying accidental complexity. Fixes: #6618
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ctx->state = (struct lavc_state){0};
}
ad_lavc, vd_lavc: return full error codes to shared decoder loop ad_lavc and vd_lavc use the lavc_process() helper to translate the FFmpeg push/pull API to the internal filter API (which completely mismatch, even though I'm responsible for both, just fucking kill me). This interface was "slightly" too tight. It returned only a bool indicating "progress", which was not enough to handle some cases (see following commit). While we're at it, move all state into a struct. This is only a single bool, but we get the chance to add more if needed. This fixes mpv falling asleep if decoding returns an error during draining. If decoding fails when we already sent EOF, the state machine stopped making progress. This left mpv just sitting around and doing nothing. A test case can be created with: echo $RANDOM >> image.png This makes libavformat read a proper packet plus a packet of garbage. libavcodec will decode a frame, and then return an error code. The lavc_process() wrapper could not deal with this, because there was no way to differentiate between "retry" and "send new packet". Normally, it would send a new packet, so decoding would make progress anyway. If there was "progress", we couldn't just retry, because it'd retry forever. This is made worse by the fact that it tries to decode at least two frames before starting display, meaning it will "sit around and do nothing" before the picture is displayed. Change it so that on error return, "receiving" a frame is retried. This will make it return the EOF, so everything works properly. This is a high-risk change, because all these funny bullshit exceptions for hardware decoding are in the way, and I didn't retest them. For example, if hardware decoding is enabled, it keeps a list of packets, that are fed into the decoder again if hardware decoding fails, and a software fallback is performed. Another case of horrifying accidental complexity. Fixes: #6618
2019-10-24 16:40:46 +00:00
static int send_packet(struct mp_filter *da, struct demux_packet *mpkt)
{
struct priv *priv = da->priv;
AVCodecContext *avctx = priv->avctx;
// If the decoder discards the timestamp for some reason, we use the
// interpolated PTS. Initialize it so that it works for the initial
// packet as well.
if (mpkt && priv->next_pts == MP_NOPTS_VALUE)
priv->next_pts = mpkt->pts;
AVPacket pkt;
mp_set_av_packet(&pkt, mpkt, &priv->codec_timebase);
int ret = avcodec_send_packet(avctx, mpkt ? &pkt : NULL);
if (ret < 0)
MP_ERR(da, "Error decoding audio.\n");
ad_lavc, vd_lavc: return full error codes to shared decoder loop ad_lavc and vd_lavc use the lavc_process() helper to translate the FFmpeg push/pull API to the internal filter API (which completely mismatch, even though I'm responsible for both, just fucking kill me). This interface was "slightly" too tight. It returned only a bool indicating "progress", which was not enough to handle some cases (see following commit). While we're at it, move all state into a struct. This is only a single bool, but we get the chance to add more if needed. This fixes mpv falling asleep if decoding returns an error during draining. If decoding fails when we already sent EOF, the state machine stopped making progress. This left mpv just sitting around and doing nothing. A test case can be created with: echo $RANDOM >> image.png This makes libavformat read a proper packet plus a packet of garbage. libavcodec will decode a frame, and then return an error code. The lavc_process() wrapper could not deal with this, because there was no way to differentiate between "retry" and "send new packet". Normally, it would send a new packet, so decoding would make progress anyway. If there was "progress", we couldn't just retry, because it'd retry forever. This is made worse by the fact that it tries to decode at least two frames before starting display, meaning it will "sit around and do nothing" before the picture is displayed. Change it so that on error return, "receiving" a frame is retried. This will make it return the EOF, so everything works properly. This is a high-risk change, because all these funny bullshit exceptions for hardware decoding are in the way, and I didn't retest them. For example, if hardware decoding is enabled, it keeps a list of packets, that are fed into the decoder again if hardware decoding fails, and a software fallback is performed. Another case of horrifying accidental complexity. Fixes: #6618
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return ret;
}
ad_lavc, vd_lavc: return full error codes to shared decoder loop ad_lavc and vd_lavc use the lavc_process() helper to translate the FFmpeg push/pull API to the internal filter API (which completely mismatch, even though I'm responsible for both, just fucking kill me). This interface was "slightly" too tight. It returned only a bool indicating "progress", which was not enough to handle some cases (see following commit). While we're at it, move all state into a struct. This is only a single bool, but we get the chance to add more if needed. This fixes mpv falling asleep if decoding returns an error during draining. If decoding fails when we already sent EOF, the state machine stopped making progress. This left mpv just sitting around and doing nothing. A test case can be created with: echo $RANDOM >> image.png This makes libavformat read a proper packet plus a packet of garbage. libavcodec will decode a frame, and then return an error code. The lavc_process() wrapper could not deal with this, because there was no way to differentiate between "retry" and "send new packet". Normally, it would send a new packet, so decoding would make progress anyway. If there was "progress", we couldn't just retry, because it'd retry forever. This is made worse by the fact that it tries to decode at least two frames before starting display, meaning it will "sit around and do nothing" before the picture is displayed. Change it so that on error return, "receiving" a frame is retried. This will make it return the EOF, so everything works properly. This is a high-risk change, because all these funny bullshit exceptions for hardware decoding are in the way, and I didn't retest them. For example, if hardware decoding is enabled, it keeps a list of packets, that are fed into the decoder again if hardware decoding fails, and a software fallback is performed. Another case of horrifying accidental complexity. Fixes: #6618
2019-10-24 16:40:46 +00:00
static int receive_frame(struct mp_filter *da, struct mp_frame *out)
{
struct priv *priv = da->priv;
AVCodecContext *avctx = priv->avctx;
int ret = avcodec_receive_frame(avctx, priv->avframe);
if (ret == AVERROR_EOF) {
// If flushing was initialized earlier and has ended now, make it start
// over in case we get new packets at some point in the future.
// (Dont' reset the filter itself, we want to keep other state.)
avcodec_flush_buffers(priv->avctx);
ad_lavc, vd_lavc: return full error codes to shared decoder loop ad_lavc and vd_lavc use the lavc_process() helper to translate the FFmpeg push/pull API to the internal filter API (which completely mismatch, even though I'm responsible for both, just fucking kill me). This interface was "slightly" too tight. It returned only a bool indicating "progress", which was not enough to handle some cases (see following commit). While we're at it, move all state into a struct. This is only a single bool, but we get the chance to add more if needed. This fixes mpv falling asleep if decoding returns an error during draining. If decoding fails when we already sent EOF, the state machine stopped making progress. This left mpv just sitting around and doing nothing. A test case can be created with: echo $RANDOM >> image.png This makes libavformat read a proper packet plus a packet of garbage. libavcodec will decode a frame, and then return an error code. The lavc_process() wrapper could not deal with this, because there was no way to differentiate between "retry" and "send new packet". Normally, it would send a new packet, so decoding would make progress anyway. If there was "progress", we couldn't just retry, because it'd retry forever. This is made worse by the fact that it tries to decode at least two frames before starting display, meaning it will "sit around and do nothing" before the picture is displayed. Change it so that on error return, "receiving" a frame is retried. This will make it return the EOF, so everything works properly. This is a high-risk change, because all these funny bullshit exceptions for hardware decoding are in the way, and I didn't retest them. For example, if hardware decoding is enabled, it keeps a list of packets, that are fed into the decoder again if hardware decoding fails, and a software fallback is performed. Another case of horrifying accidental complexity. Fixes: #6618
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return ret;
} else if (ret < 0 && ret != AVERROR(EAGAIN)) {
MP_ERR(da, "Error decoding audio.\n");
}
if (priv->avframe->flags & AV_FRAME_FLAG_DISCARD)
av_frame_unref(priv->avframe);
if (!priv->avframe->buf[0])
ad_lavc, vd_lavc: return full error codes to shared decoder loop ad_lavc and vd_lavc use the lavc_process() helper to translate the FFmpeg push/pull API to the internal filter API (which completely mismatch, even though I'm responsible for both, just fucking kill me). This interface was "slightly" too tight. It returned only a bool indicating "progress", which was not enough to handle some cases (see following commit). While we're at it, move all state into a struct. This is only a single bool, but we get the chance to add more if needed. This fixes mpv falling asleep if decoding returns an error during draining. If decoding fails when we already sent EOF, the state machine stopped making progress. This left mpv just sitting around and doing nothing. A test case can be created with: echo $RANDOM >> image.png This makes libavformat read a proper packet plus a packet of garbage. libavcodec will decode a frame, and then return an error code. The lavc_process() wrapper could not deal with this, because there was no way to differentiate between "retry" and "send new packet". Normally, it would send a new packet, so decoding would make progress anyway. If there was "progress", we couldn't just retry, because it'd retry forever. This is made worse by the fact that it tries to decode at least two frames before starting display, meaning it will "sit around and do nothing" before the picture is displayed. Change it so that on error return, "receiving" a frame is retried. This will make it return the EOF, so everything works properly. This is a high-risk change, because all these funny bullshit exceptions for hardware decoding are in the way, and I didn't retest them. For example, if hardware decoding is enabled, it keeps a list of packets, that are fed into the decoder again if hardware decoding fails, and a software fallback is performed. Another case of horrifying accidental complexity. Fixes: #6618
2019-10-24 16:40:46 +00:00
return ret;
double out_pts = mp_pts_from_av(priv->avframe->pts, &priv->codec_timebase);
audio: introduce a new type to hold audio frames This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
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struct mp_aframe *mpframe = mp_aframe_from_avframe(priv->avframe);
if (!mpframe) {
MP_ERR(da, "Converting libavcodec frame to mpv frame failed.\n");
ad_lavc, vd_lavc: return full error codes to shared decoder loop ad_lavc and vd_lavc use the lavc_process() helper to translate the FFmpeg push/pull API to the internal filter API (which completely mismatch, even though I'm responsible for both, just fucking kill me). This interface was "slightly" too tight. It returned only a bool indicating "progress", which was not enough to handle some cases (see following commit). While we're at it, move all state into a struct. This is only a single bool, but we get the chance to add more if needed. This fixes mpv falling asleep if decoding returns an error during draining. If decoding fails when we already sent EOF, the state machine stopped making progress. This left mpv just sitting around and doing nothing. A test case can be created with: echo $RANDOM >> image.png This makes libavformat read a proper packet plus a packet of garbage. libavcodec will decode a frame, and then return an error code. The lavc_process() wrapper could not deal with this, because there was no way to differentiate between "retry" and "send new packet". Normally, it would send a new packet, so decoding would make progress anyway. If there was "progress", we couldn't just retry, because it'd retry forever. This is made worse by the fact that it tries to decode at least two frames before starting display, meaning it will "sit around and do nothing" before the picture is displayed. Change it so that on error return, "receiving" a frame is retried. This will make it return the EOF, so everything works properly. This is a high-risk change, because all these funny bullshit exceptions for hardware decoding are in the way, and I didn't retest them. For example, if hardware decoding is enabled, it keeps a list of packets, that are fed into the decoder again if hardware decoding fails, and a software fallback is performed. Another case of horrifying accidental complexity. Fixes: #6618
2019-10-24 16:40:46 +00:00
return ret;
}
if (priv->force_channel_map.num)
mp_aframe_set_chmap(mpframe, &priv->force_channel_map);
audio: introduce a new type to hold audio frames This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
2017-08-16 19:00:20 +00:00
if (out_pts == MP_NOPTS_VALUE)
out_pts = priv->next_pts;
mp_aframe_set_pts(mpframe, out_pts);
audio: introduce a new type to hold audio frames This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
2017-08-16 19:00:20 +00:00
priv->next_pts = mp_aframe_end_pts(mpframe);
AVFrameSideData *sd =
av_frame_get_side_data(priv->avframe, AV_FRAME_DATA_SKIP_SAMPLES);
if (sd && sd->size >= 10) {
char *d = sd->data;
priv->skip_samples += AV_RL32(d + 0);
priv->trim_samples += AV_RL32(d + 4);
}
if (!priv->preroll_done) {
// Skip only if this isn't already handled by AV_FRAME_DATA_SKIP_SAMPLES.
if (!priv->skip_samples)
priv->skip_samples = avctx->delay;
priv->preroll_done = true;
}
audio: introduce a new type to hold audio frames This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
2017-08-16 19:00:20 +00:00
uint32_t skip = MPMIN(priv->skip_samples, mp_aframe_get_size(mpframe));
if (skip) {
audio: introduce a new type to hold audio frames This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
2017-08-16 19:00:20 +00:00
mp_aframe_skip_samples(mpframe, skip);
priv->skip_samples -= skip;
}
audio: introduce a new type to hold audio frames This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
2017-08-16 19:00:20 +00:00
uint32_t trim = MPMIN(priv->trim_samples, mp_aframe_get_size(mpframe));
if (trim) {
audio: introduce a new type to hold audio frames This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
2017-08-16 19:00:20 +00:00
mp_aframe_set_size(mpframe, mp_aframe_get_size(mpframe) - trim);
priv->trim_samples -= trim;
}
if (mp_aframe_get_size(mpframe) > 0) {
*out = MAKE_FRAME(MP_FRAME_AUDIO, mpframe);
} else {
talloc_free(mpframe);
}
av_frame_unref(priv->avframe);
ad_lavc, vd_lavc: return full error codes to shared decoder loop ad_lavc and vd_lavc use the lavc_process() helper to translate the FFmpeg push/pull API to the internal filter API (which completely mismatch, even though I'm responsible for both, just fucking kill me). This interface was "slightly" too tight. It returned only a bool indicating "progress", which was not enough to handle some cases (see following commit). While we're at it, move all state into a struct. This is only a single bool, but we get the chance to add more if needed. This fixes mpv falling asleep if decoding returns an error during draining. If decoding fails when we already sent EOF, the state machine stopped making progress. This left mpv just sitting around and doing nothing. A test case can be created with: echo $RANDOM >> image.png This makes libavformat read a proper packet plus a packet of garbage. libavcodec will decode a frame, and then return an error code. The lavc_process() wrapper could not deal with this, because there was no way to differentiate between "retry" and "send new packet". Normally, it would send a new packet, so decoding would make progress anyway. If there was "progress", we couldn't just retry, because it'd retry forever. This is made worse by the fact that it tries to decode at least two frames before starting display, meaning it will "sit around and do nothing" before the picture is displayed. Change it so that on error return, "receiving" a frame is retried. This will make it return the EOF, so everything works properly. This is a high-risk change, because all these funny bullshit exceptions for hardware decoding are in the way, and I didn't retest them. For example, if hardware decoding is enabled, it keeps a list of packets, that are fed into the decoder again if hardware decoding fails, and a software fallback is performed. Another case of horrifying accidental complexity. Fixes: #6618
2019-10-24 16:40:46 +00:00
return ret;
}
core: redo how codecs are mapped, remove codecs.conf Use codec names instead of FourCCs to identify codecs. Rewrite how codecs are selected and initialized. Now each decoder exports a list of decoders (and the codec it supports) via add_decoders(). The order matters, and the first decoder for a given decoder is preferred over the other decoders. E.g. all ad_mpg123 decoders are preferred over ad_lavc, because it comes first in the mpcodecs_ad_drivers array. Likewise, decoders within ad_lavc that are enumerated first by libavcodec (using av_codec_next()) are preferred. (This is actually critical to select h264 software decoding by default instead of vdpau. libavcodec and ffmpeg/avconv use the same method to select decoders by default, so we hope this is sane.) The codec names follow libavcodec's codec names as defined by AVCodecDescriptor.name (see libavcodec/codec_desc.c). Some decoders have names different from the canonical codec name. The AVCodecDescriptor API is relatively new, so we need a compatibility layer for older libavcodec versions for codec names that are referenced internally, and which are different from the decoder name. (Add a configure check for that, because checking versions is getting way too messy.) demux/codec_tags.c is generated from the former codecs.conf (minus "special" decoders like vdpau, and excluding the mappings that are the same as the mappings libavformat's exported RIFF tables). It contains all the mappings from FourCCs to codec name. This is needed for demux_mkv, demux_mpg, demux_avi and demux_asf. demux_lavf will set the codec as determined by libavformat, while the other demuxers have to do this on their own, using the mp_set_audio/video_codec_from_tag() functions. Note that the sh_audio/video->format members don't uniquely identify the codec anymore, and sh->codec takes over this role. Replace the --ac/--vc/--afm/--vfm with new --vd/--ad options, which provide cover the functionality of the removed switched. Note: there's no CODECS_FLAG_FLIP flag anymore. This means some obscure container/video combinations (e.g. the sample Film_200_zygo_pro.mov) are played flipped. ffplay/avplay doesn't handle this properly either, so we don't care and blame ffmeg/libav instead.
2013-02-09 14:15:19 +00:00
static void process(struct mp_filter *ad)
{
struct priv *priv = ad->priv;
ad_lavc, vd_lavc: return full error codes to shared decoder loop ad_lavc and vd_lavc use the lavc_process() helper to translate the FFmpeg push/pull API to the internal filter API (which completely mismatch, even though I'm responsible for both, just fucking kill me). This interface was "slightly" too tight. It returned only a bool indicating "progress", which was not enough to handle some cases (see following commit). While we're at it, move all state into a struct. This is only a single bool, but we get the chance to add more if needed. This fixes mpv falling asleep if decoding returns an error during draining. If decoding fails when we already sent EOF, the state machine stopped making progress. This left mpv just sitting around and doing nothing. A test case can be created with: echo $RANDOM >> image.png This makes libavformat read a proper packet plus a packet of garbage. libavcodec will decode a frame, and then return an error code. The lavc_process() wrapper could not deal with this, because there was no way to differentiate between "retry" and "send new packet". Normally, it would send a new packet, so decoding would make progress anyway. If there was "progress", we couldn't just retry, because it'd retry forever. This is made worse by the fact that it tries to decode at least two frames before starting display, meaning it will "sit around and do nothing" before the picture is displayed. Change it so that on error return, "receiving" a frame is retried. This will make it return the EOF, so everything works properly. This is a high-risk change, because all these funny bullshit exceptions for hardware decoding are in the way, and I didn't retest them. For example, if hardware decoding is enabled, it keeps a list of packets, that are fed into the decoder again if hardware decoding fails, and a software fallback is performed. Another case of horrifying accidental complexity. Fixes: #6618
2019-10-24 16:40:46 +00:00
lavc_process(ad, &priv->state, send_packet, receive_frame);
}
static const struct mp_filter_info ad_lavc_filter = {
.name = "ad_lavc",
.priv_size = sizeof(struct priv),
.process = process,
.reset = reset,
.destroy = destroy,
};
static struct mp_decoder *create(struct mp_filter *parent,
struct mp_codec_params *codec,
const char *decoder)
{
struct mp_filter *da = mp_filter_create(parent, &ad_lavc_filter);
if (!da)
return NULL;
mp_filter_add_pin(da, MP_PIN_IN, "in");
mp_filter_add_pin(da, MP_PIN_OUT, "out");
da->log = mp_log_new(da, parent->log, NULL);
struct priv *priv = da->priv;
priv->public.f = da;
if (!init(da, codec, decoder)) {
talloc_free(da);
return NULL;
}
return &priv->public;
}
core: redo how codecs are mapped, remove codecs.conf Use codec names instead of FourCCs to identify codecs. Rewrite how codecs are selected and initialized. Now each decoder exports a list of decoders (and the codec it supports) via add_decoders(). The order matters, and the first decoder for a given decoder is preferred over the other decoders. E.g. all ad_mpg123 decoders are preferred over ad_lavc, because it comes first in the mpcodecs_ad_drivers array. Likewise, decoders within ad_lavc that are enumerated first by libavcodec (using av_codec_next()) are preferred. (This is actually critical to select h264 software decoding by default instead of vdpau. libavcodec and ffmpeg/avconv use the same method to select decoders by default, so we hope this is sane.) The codec names follow libavcodec's codec names as defined by AVCodecDescriptor.name (see libavcodec/codec_desc.c). Some decoders have names different from the canonical codec name. The AVCodecDescriptor API is relatively new, so we need a compatibility layer for older libavcodec versions for codec names that are referenced internally, and which are different from the decoder name. (Add a configure check for that, because checking versions is getting way too messy.) demux/codec_tags.c is generated from the former codecs.conf (minus "special" decoders like vdpau, and excluding the mappings that are the same as the mappings libavformat's exported RIFF tables). It contains all the mappings from FourCCs to codec name. This is needed for demux_mkv, demux_mpg, demux_avi and demux_asf. demux_lavf will set the codec as determined by libavformat, while the other demuxers have to do this on their own, using the mp_set_audio/video_codec_from_tag() functions. Note that the sh_audio/video->format members don't uniquely identify the codec anymore, and sh->codec takes over this role. Replace the --ac/--vc/--afm/--vfm with new --vd/--ad options, which provide cover the functionality of the removed switched. Note: there's no CODECS_FLAG_FLIP flag anymore. This means some obscure container/video combinations (e.g. the sample Film_200_zygo_pro.mov) are played flipped. ffplay/avplay doesn't handle this properly either, so we don't care and blame ffmeg/libav instead.
2013-02-09 14:15:19 +00:00
static void add_decoders(struct mp_decoder_list *list)
{
mp_add_lavc_decoders(list, AVMEDIA_TYPE_AUDIO);
}
const struct mp_decoder_fns ad_lavc = {
.create = create,
.add_decoders = add_decoders,
};