mirror of
https://github.com/mpv-player/mpv
synced 2024-12-26 09:02:38 +00:00
cosmetics: ad_ffmpeg.c: reformat
This commit is contained in:
Uoti Urpala
parent
2c5285c151
commit
da2b884c70
@ -20,6 +20,8 @@
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#include <stdlib.h>
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#include <unistd.h>
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#include <libavcodec/avcodec.h>
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#include "config.h"
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#include "mp_msg.h"
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#include "options.h"
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@ -31,24 +33,19 @@
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static const ad_info_t info =
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{
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"FFmpeg/libavcodec audio decoders",
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"ffmpeg",
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"Nick Kurshev",
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"ffmpeg.sf.net",
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""
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"FFmpeg/libavcodec audio decoders",
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"ffmpeg",
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"Nick Kurshev",
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"ffmpeg.sf.net",
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""
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};
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LIBAD_EXTERN(ffmpeg)
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#define assert(x)
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#include "libavcodec/avcodec.h"
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static int preinit(sh_audio_t *sh)
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{
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sh->audio_out_minsize=AVCODEC_MAX_AUDIO_FRAME_SIZE;
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return 1;
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sh->audio_out_minsize = AVCODEC_MAX_AUDIO_FRAME_SIZE;
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return 1;
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}
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/* Prefer playing audio with the samplerate given in container data
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@ -56,16 +53,17 @@ static int preinit(sh_audio_t *sh)
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* from the codec, since if the codec isn't using the correct values for
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* those everything breaks anyway.
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*/
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static int setup_format(sh_audio_t *sh_audio, const AVCodecContext *lavc_context)
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static int setup_format(sh_audio_t *sh_audio,
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const AVCodecContext *lavc_context)
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{
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int sample_format = sh_audio->sample_format;
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switch (lavc_context->sample_fmt) {
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case SAMPLE_FMT_U8: sample_format = AF_FORMAT_U8; break;
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case SAMPLE_FMT_S16: sample_format = AF_FORMAT_S16_NE; break;
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case SAMPLE_FMT_S32: sample_format = AF_FORMAT_S32_NE; break;
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case SAMPLE_FMT_FLT: sample_format = AF_FORMAT_FLOAT_NE; break;
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default:
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mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n");
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case SAMPLE_FMT_U8: sample_format = AF_FORMAT_U8; break;
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case SAMPLE_FMT_S16: sample_format = AF_FORMAT_S16_NE; break;
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case SAMPLE_FMT_S32: sample_format = AF_FORMAT_S32_NE; break;
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case SAMPLE_FMT_FLT: sample_format = AF_FORMAT_FLOAT_NE; break;
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default:
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mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n");
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}
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bool broken_srate = false;
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@ -82,10 +80,10 @@ static int setup_format(sh_audio_t *sh_audio, const AVCodecContext *lavc_context
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if (lavc_context->channels != sh_audio->channels ||
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samplerate != sh_audio->samplerate ||
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sample_format != sh_audio->sample_format) {
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sh_audio->channels=lavc_context->channels;
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sh_audio->samplerate=samplerate;
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sh_audio->channels = lavc_context->channels;
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sh_audio->samplerate = samplerate;
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sh_audio->sample_format = sample_format;
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sh_audio->samplesize=af_fmt2bits(sh_audio->sample_format)/ 8;
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sh_audio->samplesize = af_fmt2bits(sh_audio->sample_format) / 8;
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if (broken_srate)
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mp_msg(MSGT_DECAUDIO, MSGL_WARN,
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"Ignoring broken container sample rate for AAC with SBR\n");
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@ -100,26 +98,28 @@ static int init(sh_audio_t *sh_audio)
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AVCodecContext *lavc_context;
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AVCodec *lavc_codec;
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mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n");
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mp_msg(MSGT_DECAUDIO, MSGL_V, "FFmpeg's libavcodec audio codec\n");
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lavc_codec = avcodec_find_decoder_by_name(sh_audio->codec->dll);
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if(!lavc_codec){
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mp_tmsg(MSGT_DECAUDIO,MSGL_ERR,"Cannot find codec '%s' in libavcodec...\n",sh_audio->codec->dll);
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return 0;
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if (!lavc_codec) {
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mp_tmsg(MSGT_DECAUDIO, MSGL_ERR,
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"Cannot find codec '%s' in libavcodec...\n",
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sh_audio->codec->dll);
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return 0;
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}
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lavc_context = avcodec_alloc_context();
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sh_audio->context=lavc_context;
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sh_audio->context = lavc_context;
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lavc_context->drc_scale = opts->drc_level;
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lavc_context->sample_rate = sh_audio->samplerate;
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lavc_context->bit_rate = sh_audio->i_bps * 8;
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if(sh_audio->wf){
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lavc_context->channels = sh_audio->wf->nChannels;
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lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec;
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lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8;
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lavc_context->block_align = sh_audio->wf->nBlockAlign;
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lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample;
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if (sh_audio->wf) {
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lavc_context->channels = sh_audio->wf->nChannels;
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lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec;
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lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8;
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lavc_context->block_align = sh_audio->wf->nBlockAlign;
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lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample;
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}
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lavc_context->request_channels = opts->audio_output_channels;
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lavc_context->codec_tag = sh_audio->format; //FOURCC
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@ -135,8 +135,8 @@ static int init(sh_audio_t *sh_audio)
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}
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// for QDM2
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if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata)
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{
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if (sh_audio->codecdata_len && sh_audio->codecdata &&
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!lavc_context->extradata) {
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lavc_context->extradata = av_malloc(sh_audio->codecdata_len +
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FF_INPUT_BUFFER_PADDING_SIZE);
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lavc_context->extradata_size = sh_audio->codecdata_len;
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@ -146,52 +146,51 @@ static int init(sh_audio_t *sh_audio)
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/* open it */
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if (avcodec_open(lavc_context, lavc_codec) < 0) {
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mp_tmsg(MSGT_DECAUDIO,MSGL_ERR, "Could not open codec.\n");
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mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Could not open codec.\n");
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return 0;
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}
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mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec \"%s\" init OK!\n", lavc_codec->name);
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mp_msg(MSGT_DECAUDIO, MSGL_V, "INFO: libavcodec \"%s\" init OK!\n",
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lavc_codec->name);
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// printf("\nFOURCC: 0x%X\n",sh_audio->format);
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if(sh_audio->format==0x3343414D){
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// MACE 3:1
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sh_audio->ds->ss_div = 2*3; // 1 samples/packet
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sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
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} else
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if(sh_audio->format==0x3643414D){
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// MACE 6:1
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sh_audio->ds->ss_div = 2*6; // 1 samples/packet
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sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
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}
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if (sh_audio->format == 0x3343414D) {
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// MACE 3:1
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sh_audio->ds->ss_div = 2 * 3; // 1 samples/packet
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sh_audio->ds->ss_mul = 2 * sh_audio->wf->nChannels; // 1 byte*ch/packet
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} else if (sh_audio->format == 0x3643414D) {
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// MACE 6:1
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sh_audio->ds->ss_div = 2 * 6; // 1 samples/packet
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sh_audio->ds->ss_mul = 2 * sh_audio->wf->nChannels; // 1 byte*ch/packet
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}
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// Decode at least 1 byte: (to get header filled)
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for (int tries = 0;;) {
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int x = decode_audio(sh_audio, sh_audio->a_buffer, 1,
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sh_audio->a_buffer_size);
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if (x > 0) {
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sh_audio->a_buffer_len = x;
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break;
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}
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if (++tries >= 5) {
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mp_msg(MSGT_DECAUDIO, MSGL_ERR,
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"ad_ffmpeg: initial decode failed\n");
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return 0;
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}
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}
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// Decode at least 1 byte: (to get header filled)
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for (int tries = 0;;) {
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int x = decode_audio(sh_audio, sh_audio->a_buffer, 1,
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sh_audio->a_buffer_size);
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if (x > 0) {
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sh_audio->a_buffer_len = x;
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break;
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}
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if (++tries >= 5) {
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mp_msg(MSGT_DECAUDIO, MSGL_ERR,
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"ad_ffmpeg: initial decode failed\n");
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return 0;
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}
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}
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sh_audio->i_bps=lavc_context->bit_rate/8;
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if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
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sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
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sh_audio->i_bps = lavc_context->bit_rate / 8;
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if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
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sh_audio->i_bps = sh_audio->wf->nAvgBytesPerSec;
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switch (lavc_context->sample_fmt) {
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case SAMPLE_FMT_U8:
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case SAMPLE_FMT_S16:
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case SAMPLE_FMT_S32:
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case SAMPLE_FMT_FLT:
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break;
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default:
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return 0;
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}
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return 1;
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switch (lavc_context->sample_fmt) {
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case SAMPLE_FMT_U8:
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case SAMPLE_FMT_S16:
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case SAMPLE_FMT_S32:
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case SAMPLE_FMT_FLT:
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break;
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default:
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return 0;
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}
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return 1;
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}
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static void uninit(sh_audio_t *sh)
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@ -199,77 +198,84 @@ static void uninit(sh_audio_t *sh)
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AVCodecContext *lavc_context = sh->context;
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if (avcodec_close(lavc_context) < 0)
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mp_tmsg(MSGT_DECVIDEO, MSGL_ERR, "Could not close codec.\n");
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mp_tmsg(MSGT_DECVIDEO, MSGL_ERR, "Could not close codec.\n");
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av_freep(&lavc_context->extradata);
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av_freep(&lavc_context);
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}
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static int control(sh_audio_t *sh,int cmd,void* arg, ...)
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static int control(sh_audio_t *sh, int cmd, void *arg, ...)
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{
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AVCodecContext *lavc_context = sh->context;
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switch(cmd){
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switch (cmd) {
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case ADCTRL_RESYNC_STREAM:
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avcodec_flush_buffers(lavc_context);
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ds_clear_parser(sh->ds);
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return CONTROL_TRUE;
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return CONTROL_TRUE;
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}
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return CONTROL_UNKNOWN;
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}
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static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
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static int decode_audio(sh_audio_t *sh_audio, unsigned char *buf, int minlen,
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int maxlen)
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{
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unsigned char *start=NULL;
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int y,len=-1;
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while(len<minlen){
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AVPacket pkt;
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int len2=maxlen;
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double pts;
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int x=ds_get_packet_pts(sh_audio->ds,&start, &pts);
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if(x<=0) {
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start = NULL;
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x = 0;
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ds_parse(sh_audio->ds, &start, &x, MP_NOPTS_VALUE, 0);
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if (x <= 0)
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break; // error
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} else {
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int in_size = x;
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int consumed = ds_parse(sh_audio->ds, &start, &x, pts, 0);
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sh_audio->ds->buffer_pos -= in_size - consumed;
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}
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av_init_packet(&pkt);
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pkt.data = start;
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pkt.size = x;
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if (pts != MP_NOPTS_VALUE) {
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sh_audio->pts = pts;
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sh_audio->pts_bytes = 0;
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}
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y=avcodec_decode_audio3(sh_audio->context,(int16_t*)buf,&len2,&pkt);
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//printf("return:%d samples_out:%d bitstream_in:%d sample_sum:%d\n", y, len2, x, len); fflush(stdout);
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// LATM may need many packets to find mux info
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if (y == AVERROR(EAGAIN))
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continue;
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if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; }
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if(!sh_audio->parser && y<x)
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sh_audio->ds->buffer_pos+=y-x; // put back data (HACK!)
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if(len2>0){
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if (((AVCodecContext *)sh_audio->context)->channels >= 5) {
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int samplesize = av_get_bits_per_sample_format(((AVCodecContext *)
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sh_audio->context)->sample_fmt) / 8;
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reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
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AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
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((AVCodecContext *)sh_audio->context)->channels,
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len2 / samplesize, samplesize);
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}
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//len=len2;break;
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if(len<0) len=len2; else len+=len2;
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buf+=len2;
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maxlen -= len2;
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sh_audio->pts_bytes += len2;
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}
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mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2);
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AVCodecContext *avctx = sh_audio->context;
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if (setup_format(sh_audio, sh_audio->context))
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unsigned char *start = NULL;
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int y, len = -1;
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while (len < minlen) {
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AVPacket pkt;
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int len2 = maxlen;
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double pts;
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int x = ds_get_packet_pts(sh_audio->ds, &start, &pts);
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if (x <= 0) {
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start = NULL;
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x = 0;
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ds_parse(sh_audio->ds, &start, &x, MP_NOPTS_VALUE, 0);
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if (x <= 0)
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break; // error
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} else {
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int in_size = x;
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int consumed = ds_parse(sh_audio->ds, &start, &x, pts, 0);
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sh_audio->ds->buffer_pos -= in_size - consumed;
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}
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av_init_packet(&pkt);
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pkt.data = start;
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pkt.size = x;
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if (pts != MP_NOPTS_VALUE) {
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sh_audio->pts = pts;
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sh_audio->pts_bytes = 0;
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}
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y = avcodec_decode_audio3(avctx, (int16_t *)buf, &len2, &pkt);
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// LATM may need many packets to find mux info
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if (y == AVERROR(EAGAIN))
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continue;
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if (y < 0) {
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mp_msg(MSGT_DECAUDIO, MSGL_V, "lavc_audio: error\n");
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break;
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}
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if (!sh_audio->parser && y < x)
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sh_audio->ds->buffer_pos += y - x; // put back data (HACK!)
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if (len2 > 0) {
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if (avctx->channels >= 5) {
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int samplesize = av_get_bits_per_sample_format(
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avctx->sample_fmt) / 8;
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reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
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AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
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avctx->channels,
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len2 / samplesize, samplesize);
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}
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if (len < 0)
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len = len2;
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else
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len += len2;
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buf += len2;
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maxlen -= len2;
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sh_audio->pts_bytes += len2;
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}
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mp_dbg(MSGT_DECAUDIO, MSGL_DBG2, "Decoded %d -> %d \n", y, len2);
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if (setup_format(sh_audio, avctx))
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break;
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}
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return len;
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return len;
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}
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