ffplay: add -af option
Based on a patch by Stefano Sabatini <stefasab@gmail.com>: http://ffmpeg.org/pipermail/ffmpeg-devel/2013-February/138452.html Signed-off-by: Marton Balint <cus@passwd.hu>
This commit is contained in:
Marton Balint
parent
738487f8db
commit
e96175ad7b
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@ -8,6 +8,7 @@ version <next>:
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or vice versa
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- support for Monkey's Audio versions from 3.93
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- perms and aperms filters
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- audio filtering support in ffplay
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version 1.2:
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@ -84,6 +84,12 @@ output. In the filter graph, the input is associated to the label
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ffmpeg-filters manual for more information about the filtergraph
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syntax.
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@item -af @var{filter_graph}
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@var{filter_graph} is a description of the filter graph to apply to
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the input audio.
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Use the option "-filters" to show all the available filters (including
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sources and sinks).
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@item -i @var{input_file}
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Read @var{input_file}.
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@end table
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169
ffplay.c
169
ffplay.c
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@ -191,7 +191,11 @@ typedef struct VideoState {
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AVPacket audio_pkt_temp;
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AVPacket audio_pkt;
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int audio_pkt_temp_serial;
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int audio_last_serial;
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struct AudioParams audio_src;
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#if CONFIG_AVFILTER
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struct AudioParams audio_filter_src;
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#endif
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struct AudioParams audio_tgt;
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struct SwrContext *swr_ctx;
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double audio_current_pts;
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@ -253,6 +257,9 @@ typedef struct VideoState {
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#if CONFIG_AVFILTER
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AVFilterContext *in_video_filter; // the first filter in the video chain
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AVFilterContext *out_video_filter; // the last filter in the video chain
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AVFilterContext *in_audio_filter; // the first filter in the audio chain
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AVFilterContext *out_audio_filter; // the last filter in the audio chain
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AVFilterGraph *agraph; // audio filter graph
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#endif
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int last_video_stream, last_audio_stream, last_subtitle_stream;
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@ -309,6 +316,7 @@ static int64_t cursor_last_shown;
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static int cursor_hidden = 0;
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#if CONFIG_AVFILTER
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static char *vfilters = NULL;
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static char *afilters = NULL;
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#endif
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/* current context */
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@ -322,6 +330,26 @@ static AVPacket flush_pkt;
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static SDL_Surface *screen;
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static inline
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int cmp_audio_fmts(enum AVSampleFormat fmt1, int64_t channel_count1,
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enum AVSampleFormat fmt2, int64_t channel_count2)
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{
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/* If channel count == 1, planar and non-planar formats are the same */
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if (channel_count1 == 1 && channel_count2 == 1)
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return av_get_packed_sample_fmt(fmt1) != av_get_packed_sample_fmt(fmt2);
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else
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return channel_count1 != channel_count2 || fmt1 != fmt2;
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}
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static inline
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int64_t get_valid_channel_layout(int64_t channel_layout, int channels)
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{
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if (channel_layout && av_get_channel_layout_nb_channels(channel_layout) == channels)
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return channel_layout;
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else
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return 0;
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}
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static int packet_queue_put(PacketQueue *q, AVPacket *pkt);
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static int packet_queue_put_private(PacketQueue *q, AVPacket *pkt)
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@ -1781,6 +1809,71 @@ fail:
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return ret;
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}
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static int configure_audio_filters(VideoState *is, const char *afilters, int force_output_format)
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{
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static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_S16, PIX_FMT_NONE };
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int sample_rates[2] = { 0, -1 };
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int64_t channel_layouts[2] = { 0, -1 };
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int channels[2] = { 0, -1 };
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AVFilterContext *filt_asrc = NULL, *filt_asink = NULL;
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char asrc_args[256];
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AVABufferSinkParams *asink_params = NULL;
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int ret;
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avfilter_graph_free(&is->agraph);
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if (!(is->agraph = avfilter_graph_alloc()))
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return AVERROR(ENOMEM);
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ret = snprintf(asrc_args, sizeof(asrc_args),
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"sample_rate=%d:sample_fmt=%s:channels=%d",
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is->audio_filter_src.freq, av_get_sample_fmt_name(is->audio_filter_src.fmt),
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is->audio_filter_src.channels);
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if (is->audio_filter_src.channel_layout)
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snprintf(asrc_args + ret, sizeof(asrc_args) - ret,
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":channel_layout=0x%"PRIx64, is->audio_filter_src.channel_layout);
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ret = avfilter_graph_create_filter(&filt_asrc,
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avfilter_get_by_name("abuffer"), "ffplay_abuffer",
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asrc_args, NULL, is->agraph);
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if (ret < 0)
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goto end;
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if (!(asink_params = av_abuffersink_params_alloc())) {
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ret = AVERROR(ENOMEM);
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goto end;
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}
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asink_params->sample_fmts = sample_fmts;
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asink_params->all_channel_counts = 1;
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if (force_output_format) {
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channel_layouts[0] = is->audio_tgt.channel_layout;
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asink_params->channel_layouts = channel_layouts;
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asink_params->all_channel_counts = 0;
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channels[0] = is->audio_tgt.channels;
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asink_params->channel_counts = channels;
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asink_params->all_channel_counts = 0;
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sample_rates[0] = is->audio_tgt.freq;
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asink_params->sample_rates = sample_rates;
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}
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ret = avfilter_graph_create_filter(&filt_asink,
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avfilter_get_by_name("abuffersink"), "ffplay_abuffersink",
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NULL, asink_params, is->agraph);
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if (ret < 0)
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goto end;
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if ((ret = configure_filtergraph(is->agraph, afilters, filt_asrc, filt_asink)) < 0)
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goto end;
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is->in_audio_filter = filt_asrc;
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is->out_audio_filter = filt_asink;
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end:
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av_freep(&asink_params);
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if (ret < 0)
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avfilter_graph_free(&is->agraph);
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return ret;
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}
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#endif /* CONFIG_AVFILTER */
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static int video_thread(void *arg)
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@ -2056,6 +2149,7 @@ static int audio_decode_frame(VideoState *is)
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int new_packet = 0;
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int flush_complete = 0;
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int wanted_nb_samples;
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AVRational tb;
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for (;;) {
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/* NOTE: the audio packet can contain several frames */
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@ -2098,6 +2192,49 @@ static int audio_decode_frame(VideoState *is)
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is->frame->pts = av_rescale_q(pkt_temp->pts, is->audio_st->time_base, dec->time_base);
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if (pkt_temp->pts != AV_NOPTS_VALUE)
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pkt_temp->pts += (double) is->frame->nb_samples / is->frame->sample_rate / av_q2d(is->audio_st->time_base);
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tb = dec->time_base;
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#if CONFIG_AVFILTER
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{
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int ret;
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int reconfigure;
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dec_channel_layout = get_valid_channel_layout(is->frame->channel_layout, av_frame_get_channels(is->frame));
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reconfigure =
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cmp_audio_fmts(is->audio_filter_src.fmt, is->audio_filter_src.channels,
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is->frame->format, av_frame_get_channels(is->frame)) ||
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is->audio_filter_src.channel_layout != dec_channel_layout ||
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is->audio_filter_src.freq != is->frame->sample_rate ||
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is->audio_pkt_temp_serial != is->audio_last_serial;
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if (reconfigure) {
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char buf1[1024], buf2[1024];
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av_get_channel_layout_string(buf1, sizeof(buf1), -1, is->audio_filter_src.channel_layout);
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av_get_channel_layout_string(buf2, sizeof(buf2), -1, dec_channel_layout);
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av_log(NULL, AV_LOG_DEBUG,
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"Audio frame changed from rate:%d ch:%d fmt:%s layout:%s serial:%d to rate:%d ch:%d fmt:%s layout:%s serial:%d\n",
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is->audio_filter_src.freq, is->audio_filter_src.channels, av_get_sample_fmt_name(is->audio_filter_src.fmt), buf1, is->audio_last_serial,
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is->frame->sample_rate, av_frame_get_channels(is->frame), av_get_sample_fmt_name(is->frame->format), buf2, is->audio_pkt_temp_serial);
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is->audio_filter_src.fmt = is->frame->format;
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is->audio_filter_src.channels = av_frame_get_channels(is->frame);
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is->audio_filter_src.channel_layout = dec_channel_layout;
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is->audio_filter_src.freq = is->frame->sample_rate;
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is->audio_last_serial = is->audio_pkt_temp_serial;
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if ((ret = configure_audio_filters(is, afilters, 1)) < 0)
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return ret;
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}
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if ((ret = av_buffersrc_add_frame(is->in_audio_filter, is->frame)) < 0)
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return ret;
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av_frame_unref(is->frame);
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if ((ret = av_buffersink_get_frame_flags(is->out_audio_filter, is->frame, 0)) < 0)
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return ret;
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tb = is->out_audio_filter->inputs[0]->time_base;
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}
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#endif
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data_size = av_samples_get_buffer_size(NULL, av_frame_get_channels(is->frame),
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is->frame->nb_samples,
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@ -2164,7 +2301,7 @@ static int audio_decode_frame(VideoState *is)
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audio_clock0 = is->audio_clock;
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/* update the audio clock with the pts */
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if (is->frame->pts != AV_NOPTS_VALUE) {
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is->audio_clock = is->frame->pts * av_q2d(dec->time_base) + (double) is->frame->nb_samples / is->frame->sample_rate;
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is->audio_clock = is->frame->pts * av_q2d(tb) + (double) is->frame->nb_samples / is->frame->sample_rate;
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is->audio_clock_serial = is->audio_pkt_temp_serial;
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}
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#ifdef DEBUG
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@ -2308,6 +2445,8 @@ static int stream_component_open(VideoState *is, int stream_index)
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const char *forced_codec_name = NULL;
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AVDictionary *opts;
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AVDictionaryEntry *t = NULL;
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int sample_rate, nb_channels;
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int64_t channel_layout;
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int ret;
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if (stream_index < 0 || stream_index >= ic->nb_streams)
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@ -2363,8 +2502,29 @@ static int stream_component_open(VideoState *is, int stream_index)
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ic->streams[stream_index]->discard = AVDISCARD_DEFAULT;
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switch (avctx->codec_type) {
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case AVMEDIA_TYPE_AUDIO:
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#if CONFIG_AVFILTER
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{
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AVFilterLink *link;
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is->audio_filter_src.freq = avctx->sample_rate;
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is->audio_filter_src.channels = avctx->channels;
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is->audio_filter_src.channel_layout = get_valid_channel_layout(avctx->channel_layout, avctx->channels);
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is->audio_filter_src.fmt = avctx->sample_fmt;
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if ((ret = configure_audio_filters(is, afilters, 0)) < 0)
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return ret;
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link = is->out_audio_filter->inputs[0];
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sample_rate = link->sample_rate;
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nb_channels = link->channels;
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channel_layout = link->channel_layout;
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}
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#else
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sample_rate = avctx->sample_rate;
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nb_channels = avctx->channels;
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channel_layout = avctx->channel_layout;
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#endif
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/* prepare audio output */
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if ((ret = audio_open(is, avctx->channel_layout, avctx->channels, avctx->sample_rate, &is->audio_tgt)) < 0)
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if ((ret = audio_open(is, channel_layout, nb_channels, sample_rate, &is->audio_tgt)) < 0)
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return ret;
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is->audio_hw_buf_size = ret;
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is->audio_src = is->audio_tgt;
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@ -2436,6 +2596,9 @@ static void stream_component_close(VideoState *is, int stream_index)
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is->rdft = NULL;
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is->rdft_bits = 0;
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}
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#if CONFIG_AVFILTER
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avfilter_graph_free(&is->agraph);
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#endif
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break;
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case AVMEDIA_TYPE_VIDEO:
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packet_queue_abort(&is->videoq);
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@ -2825,6 +2988,7 @@ static VideoState *stream_open(const char *filename, AVInputFormat *iformat)
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is->video_current_pts_drift = is->audio_current_pts_drift;
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is->audio_clock_serial = -1;
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is->video_clock_serial = -1;
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is->audio_last_serial = -1;
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is->av_sync_type = av_sync_type;
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is->read_tid = SDL_CreateThread(read_thread, is);
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if (!is->read_tid) {
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@ -3233,6 +3397,7 @@ static const OptionDef options[] = {
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{ "window_title", OPT_STRING | HAS_ARG, { &window_title }, "set window title", "window title" },
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#if CONFIG_AVFILTER
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{ "vf", OPT_STRING | HAS_ARG, { &vfilters }, "set video filters", "filter_graph" },
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{ "af", OPT_STRING | HAS_ARG, { &afilters }, "set audio filters", "filter_graph" },
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#endif
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{ "rdftspeed", OPT_INT | HAS_ARG| OPT_AUDIO | OPT_EXPERT, { &rdftspeed }, "rdft speed", "msecs" },
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{ "showmode", HAS_ARG, { .func_arg = opt_show_mode}, "select show mode (0 = video, 1 = waves, 2 = RDFT)", "mode" },
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