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https://github.com/mpv-player/mpv
synced 2025-02-18 05:37:04 +00:00
audio: support parameter changes (e.g. channel count) during playback
Add support for parameter changes (e.g. channel count) during playback. This makes decoding AC3 files that switch between 2 and 6 channels work reasonably well even with -channels 6 and ffac3 decoder. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@31737 b3059339-0415-0410-9bf9-f77b7e298cf2 Fix typo in error message: ACC -> AAC git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32473 b3059339-0415-0410-9bf9-f77b7e298cf2 Avoid printing AAC with SBR warning on every decode call, instead print it only after every decoder reconfiguration. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32476 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
reimar
Uoti Urpala
parent
6be6ba4094
commit
5ed772b9cd
@ -52,6 +52,44 @@ static int preinit(sh_audio_t *sh)
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return 1;
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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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{
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int broken_srate = 0;
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int samplerate = lavc_context->sample_rate;
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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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}
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if(sh_audio->wf){
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// If the decoder uses the wrong number of channels all is lost anyway.
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// sh_audio->channels=sh_audio->wf->nChannels;
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if (lavc_context->codec_id == CODEC_ID_AAC &&
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samplerate == 2*sh_audio->wf->nSamplesPerSec) {
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broken_srate = 1;
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} else if (sh_audio->wf->nSamplesPerSec)
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samplerate=sh_audio->wf->nSamplesPerSec;
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}
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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->sample_format = sample_format;
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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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return 1;
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}
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return 0;
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}
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static int init(sh_audio_t *sh_audio)
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{
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struct MPOpts *opts = sh_audio->opts;
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@ -133,32 +171,19 @@ static int init(sh_audio_t *sh_audio)
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} while (x <= 0 && tries++ < 5);
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if(x>0) sh_audio->a_buffer_len=x;
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sh_audio->channels=lavc_context->channels;
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sh_audio->samplerate=lavc_context->sample_rate;
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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: sh_audio->sample_format = AF_FORMAT_U8; break;
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case SAMPLE_FMT_S16: sh_audio->sample_format = AF_FORMAT_S16_NE; break;
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case SAMPLE_FMT_S32: sh_audio->sample_format = AF_FORMAT_S32_NE; break;
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case SAMPLE_FMT_FLT: sh_audio->sample_format = AF_FORMAT_FLOAT_NE; break;
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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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mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n");
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return 0;
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}
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/* If the audio is AAC the container level data may be unreliable
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* because of SBR handling problems (possibly half real sample rate at
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* container level). Default AAC decoding with ad_faad has used codec-level
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* values for a long time without generating complaints so it should be OK.
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*/
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if (sh_audio->wf && lavc_context->codec_id != CODEC_ID_AAC) {
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// If the decoder uses the wrong number of channels all is lost anyway.
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// sh_audio->channels=sh_audio->wf->nChannels;
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if (sh_audio->wf->nSamplesPerSec)
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sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
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if (sh_audio->wf->nAvgBytesPerSec)
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sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
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}
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sh_audio->samplesize=af_fmt2bits(sh_audio->sample_format)/ 8;
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return 1;
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}
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@ -232,6 +257,9 @@ static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int m
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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, sh_audio->context))
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break;
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}
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return len;
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}
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@ -378,13 +378,20 @@ static int filter_n_bytes(sh_audio_t *sh, int len)
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int error = 0;
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// Decode more bytes if needed
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int old_samplerate = sh->samplerate;
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int old_channels = sh->channels;
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int old_sample_format = sh->sample_format;
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while (sh->a_buffer_len < len) {
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unsigned char *buf = sh->a_buffer + sh->a_buffer_len;
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int minlen = len - sh->a_buffer_len;
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int maxlen = sh->a_buffer_size - sh->a_buffer_len;
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int ret = sh->ad_driver->decode_audio(sh, buf, minlen, maxlen);
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if (ret <= 0) {
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error = -1;
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int format_change = sh->samplerate != old_samplerate
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|| sh->channels != old_channels
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|| sh->sample_format != old_sample_format;
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if (ret <= 0 || format_change) {
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error = format_change ? -2 : -1;
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// samples from format-changing call get discarded too
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len = sh->a_buffer_len;
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break;
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}
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@ -467,8 +474,9 @@ int decode_audio(sh_audio_t *sh_audio, int minlen)
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/* if this iteration does not fill buffer, we must have lots
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* of buffering in filters */
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huge_filter_buffer = 1;
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if (filter_n_bytes(sh_audio, declen) < 0)
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return -1;
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int res = filter_n_bytes(sh_audio, declen);
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if (res < 0)
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return res;
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}
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return 0;
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}
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101
mplayer.c
101
mplayer.c
@ -1764,49 +1764,54 @@ void reinit_audio_chain(struct MPContext *mpctx)
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struct MPOpts *opts = &mpctx->opts;
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if (!mpctx->sh_audio)
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return;
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current_module="init_audio_codec";
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mp_msg(MSGT_CPLAYER,MSGL_INFO,"==========================================================================\n");
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if(!init_best_audio_codec(mpctx->sh_audio,audio_codec_list,audio_fm_list)){
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goto init_error;
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if (!(mpctx->initialized_flags & INITIALIZED_ACODEC)) {
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current_module="init_audio_codec";
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mp_msg(MSGT_CPLAYER,MSGL_INFO,"==========================================================================\n");
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if(!init_best_audio_codec(mpctx->sh_audio,audio_codec_list,audio_fm_list)){
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goto init_error;
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}
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mpctx->initialized_flags|=INITIALIZED_ACODEC;
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mp_msg(MSGT_CPLAYER,MSGL_INFO,"==========================================================================\n");
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}
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mpctx->initialized_flags|=INITIALIZED_ACODEC;
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mp_msg(MSGT_CPLAYER,MSGL_INFO,"==========================================================================\n");
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current_module="af_preinit";
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ao_data.samplerate=force_srate;
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ao_data.channels=0;
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ao_data.format=audio_output_format;
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// first init to detect best values
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if(!init_audio_filters(mpctx->sh_audio, // preliminary init
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// input:
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mpctx->sh_audio->samplerate,
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// output:
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&ao_data.samplerate, &ao_data.channels, &ao_data.format)){
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mp_tmsg(MSGT_CPLAYER,MSGL_ERR, "Error at audio filter chain "
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"pre-init!\n");
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exit_player(mpctx, EXIT_ERROR);
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if (!(mpctx->initialized_flags & INITIALIZED_AO)) {
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current_module="af_preinit";
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ao_data.samplerate=force_srate;
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ao_data.channels=0;
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ao_data.format=audio_output_format;
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// first init to detect best values
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if(!init_audio_filters(mpctx->sh_audio, // preliminary init
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// input:
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mpctx->sh_audio->samplerate,
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// output:
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&ao_data.samplerate, &ao_data.channels, &ao_data.format)){
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mp_tmsg(MSGT_CPLAYER,MSGL_ERR, "Error at audio filter chain "
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"pre-init!\n");
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exit_player(mpctx, EXIT_ERROR);
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}
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current_module="ao2_init";
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mpctx->audio_out = init_best_audio_out(opts->audio_driver_list,
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0, // plugin flag
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ao_data.samplerate,
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ao_data.channels,
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ao_data.format, 0);
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if(!mpctx->audio_out){
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mp_tmsg(MSGT_CPLAYER,MSGL_ERR,"Could not open/initialize audio device -> no sound.\n");
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goto init_error;
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}
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mpctx->initialized_flags|=INITIALIZED_AO;
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mp_msg(MSGT_CPLAYER,MSGL_INFO,"AO: [%s] %dHz %dch %s (%d bytes per sample)\n",
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mpctx->audio_out->info->short_name,
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ao_data.samplerate, ao_data.channels,
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af_fmt2str_short(ao_data.format),
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af_fmt2bits(ao_data.format)/8 );
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mp_msg(MSGT_CPLAYER,MSGL_V,"AO: Description: %s\nAO: Author: %s\n",
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mpctx->audio_out->info->name, mpctx->audio_out->info->author);
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if(strlen(mpctx->audio_out->info->comment) > 0)
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mp_msg(MSGT_CPLAYER,MSGL_V,"AO: Comment: %s\n", mpctx->audio_out->info->comment);
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}
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current_module="ao2_init";
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mpctx->audio_out = init_best_audio_out(opts->audio_driver_list,
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0, // plugin flag
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ao_data.samplerate,
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ao_data.channels,
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ao_data.format, 0);
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if(!mpctx->audio_out){
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mp_tmsg(MSGT_CPLAYER,MSGL_ERR,"Could not open/initialize audio device -> no sound.\n");
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goto init_error;
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}
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mpctx->initialized_flags|=INITIALIZED_AO;
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mp_msg(MSGT_CPLAYER,MSGL_INFO,"AO: [%s] %dHz %dch %s (%d bytes per sample)\n",
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mpctx->audio_out->info->short_name,
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ao_data.samplerate, ao_data.channels,
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af_fmt2str_short(ao_data.format),
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af_fmt2bits(ao_data.format)/8 );
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mp_msg(MSGT_CPLAYER,MSGL_V,"AO: Description: %s\nAO: Author: %s\n",
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mpctx->audio_out->info->name, mpctx->audio_out->info->author);
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if(strlen(mpctx->audio_out->info->comment) > 0)
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mp_msg(MSGT_CPLAYER,MSGL_V,"AO: Comment: %s\n", mpctx->audio_out->info->comment);
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// init audio filters:
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current_module="af_init";
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if(!build_afilter_chain(mpctx, mpctx->sh_audio, &ao_data)) {
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@ -2167,6 +2172,7 @@ static int fill_audio_out_buffers(struct MPContext *mpctx)
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int playsize;
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int playflags=0;
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int audio_eof=0;
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bool format_change = false;
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sh_audio_t * const sh_audio = mpctx->sh_audio;
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current_module="play_audio";
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@ -2192,12 +2198,16 @@ static int fill_audio_out_buffers(struct MPContext *mpctx)
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// Fill buffer if needed:
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current_module="decode_audio";
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t = GetTimer();
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if (decode_audio(sh_audio, playsize) < 0) // EOF or error
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if (mpctx->d_audio->eof) {
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int res = decode_audio(sh_audio, playsize);
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if (res < 0) { // EOF, error or format change
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if (res == -2)
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format_change = true;
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else if (mpctx->d_audio->eof) {
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audio_eof = 1;
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if (sh_audio->a_out_buffer_len == 0)
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return 0;
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}
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}
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t = GetTimer() - t;
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tt = t*0.000001f; audio_time_usage+=tt;
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if (playsize > sh_audio->a_out_buffer_len) {
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@ -2230,6 +2240,15 @@ static int fill_audio_out_buffers(struct MPContext *mpctx)
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sh_audio->a_out_buffer_len = 0;
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}
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/* The format change isn't handled too gracefully. A more precise
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* implementation would require draining buffered old-format audio
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* while displaying video, then doing the output format switch.
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*/
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if (format_change) {
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uninit_player(mpctx, INITIALIZED_AO);
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reinit_audio_chain(mpctx);
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}
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return 1;
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}
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