[FFmpeg-devel] [PATCHv3] ffplay: add -af option

Stefano Sabatini stefasab at gmail.com
Fri Mar 15 00:29:19 CET 2013


On date Wednesday 2013-03-13 22:35:12 +0100, Marton Balint encoded:
> Updated for refcounted frames.
> 
> Based on a patch by Stefano Sabatini <stefasab at gmail.com>:
> http://ffmpeg.org/pipermail/ffmpeg-devel/2013-February/138452.html
> 
> Signed-off-by: Marton Balint <cus at passwd.hu>
> ---
>  doc/ffplay.texi |    6 ++
>  ffplay.c        |  168 ++++++++++++++++++++++++++++++++++++++++++++++++++++++-
>  2 files changed, 172 insertions(+), 2 deletions(-)
> 
> diff --git a/doc/ffplay.texi b/doc/ffplay.texi
> index 5f17902..ee160a0 100644
> --- a/doc/ffplay.texi
> +++ b/doc/ffplay.texi
> @@ -84,6 +84,12 @@ output. In the filter graph, the input is associated to the label
>  ffmpeg-filters manual for more information about the filtergraph
>  syntax.
>  
> + at item -af @var{filter_graph}
> + at var{filter_graph} is a description of the filter graph to apply to
> +the input audio.
> +Use the option "-filters" to show all the available filters (including
> +sources and sinks).
> +
>  @item -i @var{input_file}
>  Read @var{input_file}.
>  @end table
> diff --git a/ffplay.c b/ffplay.c
> index 8adac1c..d07231c 100644
> --- a/ffplay.c
> +++ b/ffplay.c
> @@ -191,7 +191,11 @@ typedef struct VideoState {
>      AVPacket audio_pkt_temp;
>      AVPacket audio_pkt;
>      int audio_pkt_temp_serial;
> +    int audio_last_serial;
>      struct AudioParams audio_src;
> +#if CONFIG_AVFILTER
> +    struct AudioParams audio_filter_src;
> +#endif
>      struct AudioParams audio_tgt;
>      struct SwrContext *swr_ctx;
>      double audio_current_pts;
> @@ -253,6 +257,9 @@ typedef struct VideoState {
>  #if CONFIG_AVFILTER
>      AVFilterContext *in_video_filter;   // the first filter in the video chain
>      AVFilterContext *out_video_filter;  // the last filter in the video chain
> +    AVFilterContext *in_audio_filter;   ///<the first filter in the audio chain
> +    AVFilterContext *out_audio_filter;  ///<the last filter in the audio chain
> +    AVFilterGraph *agraph;              ///<audio filter graph
>  #endif
>  
>      int last_video_stream, last_audio_stream, last_subtitle_stream;
> @@ -309,6 +316,7 @@ static int64_t cursor_last_shown;
>  static int cursor_hidden = 0;
>  #if CONFIG_AVFILTER
>  static char *vfilters = NULL;
> +static char *afilters = NULL;
>  #endif
>  
>  /* current context */
> @@ -322,6 +330,24 @@ static AVPacket flush_pkt;
>  
>  static SDL_Surface *screen;
>  
> +static inline
> +int cmp_audio_fmts(enum AVSampleFormat fmt1, int64_t channel_count1,
> +                   enum AVSampleFormat fmt2, int64_t channel_count2)
> +{

Nit: a note about why this is necessary may be useful to the reader. I
suggest:

// in case channel_count == 1, consider formats with the same sample
// format equivalent, e.g. between mono DBL and DBLP


> +    if (channel_count1 == 1 && channel_count2 == 1)
> +        return av_get_packed_sample_fmt(fmt1) != av_get_packed_sample_fmt(fmt2);
> +    else
> +        return channel_count1 != channel_count2 || fmt1 != fmt2;


> +}
> +
> +static inline
> +int64_t get_valid_channel_layout(int64_t channel_layout, int channels) {

nit+: { on a separate line

> +    if (channel_layout && av_get_channel_layout_nb_channels(channel_layout) == channels)
> +        return channel_layout;
> +    else
> +        return 0;
> +}
> +
>  static int packet_queue_put(PacketQueue *q, AVPacket *pkt);
>  
>  static int packet_queue_put_private(PacketQueue *q, AVPacket *pkt)
> @@ -1781,6 +1807,71 @@ fail:
>      return ret;
>  }
>  
> +static int configure_audio_filters(VideoState *is, const char *afilters, int force_output_format)
> +{
> +    static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_S16, PIX_FMT_NONE };
> +    int sample_rates[2] = { 0, -1 };
> +    int64_t channel_layouts[2] = { 0, -1 };
> +    int channels[2] = { 0, -1 };
> +    AVFilterContext *filt_asrc = NULL, *filt_asink = NULL;
> +    char abuffer_args[256];
> +    AVABufferSinkParams *abuffersink_params = NULL;

nit: asink_params or filt_asink for consistency

> +    int ret;
> +
> +    avfilter_graph_free(&is->agraph);
> +    if (!(is->agraph = avfilter_graph_alloc()))
> +        return AVERROR(ENOMEM);
> +
> +    ret = snprintf(abuffer_args, sizeof(abuffer_args),
> +                   "sample_rate=%d:sample_fmt=%s:channels=%d",
> +                   is->audio_filter_src.freq, av_get_sample_fmt_name(is->audio_filter_src.fmt),
> +                   is->audio_filter_src.channels);
> +    if (is->audio_filter_src.channel_layout)
> +        snprintf(abuffer_args + ret, sizeof(abuffer_args) - ret,
> +                 ":channel_layout=0x%"PRIx64,  is->audio_filter_src.channel_layout);
> +
> +    ret = avfilter_graph_create_filter(&filt_asrc,
> +                                       avfilter_get_by_name("abuffer"), "ffplay_abuffer",
> +                                       abuffer_args, NULL, is->agraph);
> +    if (ret < 0)
> +        goto fail;
> +
> +    if (!(abuffersink_params = av_abuffersink_params_alloc())) {
> +        ret = AVERROR(ENOMEM);
> +        goto fail;
> +    }
> +    abuffersink_params->sample_fmts = sample_fmts;
> +
> +    abuffersink_params->all_channel_counts = 1;
> +    if (force_output_format) {
> +        channel_layouts[0] = is->audio_tgt.channel_layout;
> +        abuffersink_params->channel_layouts = channel_layouts;
> +        abuffersink_params->all_channel_counts = 0;
> +        channels[0] = is->audio_tgt.channels;
> +        abuffersink_params->channel_counts = channels;
> +        abuffersink_params->all_channel_counts = 0;
> +        sample_rates[0] = is->audio_tgt.freq;
> +        abuffersink_params->sample_rates = sample_rates;
> +    }
> +
> +    ret = avfilter_graph_create_filter(&filt_asink,
> +                                       avfilter_get_by_name("abuffersink"), "ffplay_abuffersink",
> +                                       NULL, abuffersink_params, is->agraph);
> +    if (ret < 0)
> +        goto fail;
> +
> +    if ((ret = configure_filtergraph(is->agraph, afilters, filt_asrc, filt_asink)) < 0)
> +        goto fail;
> +
> +    is->in_audio_filter  = filt_asrc;
> +    is->out_audio_filter = filt_asink;
> +
> +fail:

nit: "fail" -> "end" since this is executed even when there is no failure

> +    av_freep(&abuffersink_params);
> +    if (ret < 0)
> +        avfilter_graph_free(&is->agraph);
> +    return ret;
> +}
>  #endif  /* CONFIG_AVFILTER */
>  
>  static int video_thread(void *arg)
> @@ -2056,6 +2147,7 @@ static int audio_decode_frame(VideoState *is)
>      int new_packet = 0;
>      int flush_complete = 0;
>      int wanted_nb_samples;
> +    AVRational tb;
>  
>      for (;;) {
>          /* NOTE: the audio packet can contain several frames */
> @@ -2098,6 +2190,50 @@ static int audio_decode_frame(VideoState *is)
>                  is->frame->pts = av_rescale_q(pkt_temp->pts, is->audio_st->time_base, dec->time_base);
>              if (pkt_temp->pts != AV_NOPTS_VALUE)
>                  pkt_temp->pts += (double) is->frame->nb_samples / is->frame->sample_rate / av_q2d(is->audio_st->time_base);
> +            tb = dec->time_base;
> +
> +#if CONFIG_AVFILTER
> +            {
> +                int ret;
> +                int reconfigure;
> +
> +                dec_channel_layout = get_valid_channel_layout(is->frame->channel_layout, av_frame_get_channels(is->frame));
> +
> +                reconfigure =
> +                    cmp_audio_fmts(is->audio_filter_src.fmt, is->audio_filter_src.channels,
> +                                   is->frame->format, av_frame_get_channels(is->frame))    ||
> +                    is->audio_filter_src.channel_layout != dec_channel_layout ||
> +                    is->audio_filter_src.freq           != is->frame->sample_rate ||
> +                    is->audio_pkt_temp_serial           != is->audio_last_serial;
> +
> +                if (reconfigure) {
> +                    char buf1[1024], buf2[1024];
> +                    av_get_channel_layout_string(buf1, sizeof(buf1), -1, is->audio_filter_src.channel_layout);
> +                    av_get_channel_layout_string(buf2, sizeof(buf2), -1, dec_channel_layout);
> +                    av_log(NULL, AV_LOG_DEBUG,
> +                           "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",
> +                           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,
> +                           is->frame->sample_rate, av_frame_get_channels(is->frame), av_get_sample_fmt_name(is->frame->format), buf2, is->audio_pkt_temp_serial);
> +
> +                    is->audio_filter_src.fmt            = is->frame->format;
> +                    is->audio_filter_src.channels       = av_frame_get_channels(is->frame);
> +                    is->audio_filter_src.channel_layout = dec_channel_layout;
> +                    is->audio_filter_src.freq           = is->frame->sample_rate;
> +                    is->audio_last_serial               = is->audio_pkt_temp_serial;
> +
> +                    if ((ret = configure_audio_filters(is, afilters, 1)) < 0)
> +                        return ret;
> +                }
> +
> +                if ((ret = av_buffersrc_add_frame(is->in_audio_filter, is->frame)) < 0)
> +                    return ret;
> +                av_frame_unref(is->frame);

> +                avcodec_get_frame_defaults(is->frame);

is this required?

> +                if ((ret = av_buffersink_get_frame_flags(is->out_audio_filter, is->frame, 0)) < 0)
> +                    return ret;
> +                tb = is->out_audio_filter->inputs[0]->time_base;
> +            }
> +#endif
>  
>              data_size = av_samples_get_buffer_size(NULL, av_frame_get_channels(is->frame),
>                                                     is->frame->nb_samples,
> @@ -2164,7 +2300,7 @@ static int audio_decode_frame(VideoState *is)
>              audio_clock0 = is->audio_clock;
>              /* update the audio clock with the pts */
>              if (is->frame->pts != AV_NOPTS_VALUE) {
> -                is->audio_clock = is->frame->pts * av_q2d(dec->time_base) + (double) is->frame->nb_samples / is->frame->sample_rate;
> +                is->audio_clock = is->frame->pts * av_q2d(tb) + (double) is->frame->nb_samples / is->frame->sample_rate;
>                  is->audio_clock_serial = is->audio_pkt_temp_serial;
>              }
>  #ifdef DEBUG
> @@ -2308,6 +2444,8 @@ static int stream_component_open(VideoState *is, int stream_index)
>      const char *forced_codec_name = NULL;
>      AVDictionary *opts;
>      AVDictionaryEntry *t = NULL;
> +    int sample_rate, nb_channels;
> +    int64_t channel_layout;
>      int ret;
>  
>      if (stream_index < 0 || stream_index >= ic->nb_streams)
> @@ -2363,8 +2501,29 @@ static int stream_component_open(VideoState *is, int stream_index)
>      ic->streams[stream_index]->discard = AVDISCARD_DEFAULT;
>      switch (avctx->codec_type) {
>      case AVMEDIA_TYPE_AUDIO:
> +#if CONFIG_AVFILTER
> +        {
> +            AVFilterLink *link;
> +
> +            is->audio_filter_src.freq           = avctx->sample_rate;
> +            is->audio_filter_src.channels       = avctx->channels;
> +            is->audio_filter_src.channel_layout = get_valid_channel_layout(avctx->channel_layout, avctx->channels);
> +            is->audio_filter_src.fmt            = avctx->sample_fmt;
> +            if ((ret = configure_audio_filters(is, afilters, 0)) < 0)
> +                return ret;
> +            link = is->out_audio_filter->inputs[0];
> +            sample_rate    = link->sample_rate;
> +            nb_channels    = link->channels;
> +            channel_layout = link->channel_layout;
> +        }
> +#else
> +        sample_rate    = avctx->sample_rate;
> +        nb_channels    = avctx->channels;
> +        channel_layout = avctx->channel_layout;
> +#endif
> +
>          /* prepare audio output */
> -        if ((ret = audio_open(is, avctx->channel_layout, avctx->channels, avctx->sample_rate, &is->audio_tgt)) < 0)
> +        if ((ret = audio_open(is, channel_layout, nb_channels, sample_rate, &is->audio_tgt)) < 0)
>              return ret;
>          is->audio_hw_buf_size = ret;
>          is->audio_src = is->audio_tgt;
> @@ -2436,6 +2595,9 @@ static void stream_component_close(VideoState *is, int stream_index)
>              is->rdft = NULL;
>              is->rdft_bits = 0;
>          }
> +#if CONFIG_AVFILTER
> +        avfilter_graph_free(&is->agraph);
> +#endif
>          break;
>      case AVMEDIA_TYPE_VIDEO:
>          packet_queue_abort(&is->videoq);
> @@ -2825,6 +2987,7 @@ static VideoState *stream_open(const char *filename, AVInputFormat *iformat)
>      is->video_current_pts_drift = is->audio_current_pts_drift;
>      is->audio_clock_serial = -1;
>      is->video_clock_serial = -1;
> +    is->audio_last_serial = -1;
>      is->av_sync_type = av_sync_type;
>      is->read_tid     = SDL_CreateThread(read_thread, is);
>      if (!is->read_tid) {
> @@ -3233,6 +3396,7 @@ static const OptionDef options[] = {
>      { "window_title", OPT_STRING | HAS_ARG, { &window_title }, "set window title", "window title" },
>  #if CONFIG_AVFILTER
>      { "vf", OPT_STRING | HAS_ARG, { &vfilters }, "set video filters", "filter_graph" },
> +    { "af", OPT_STRING | HAS_ARG, { &afilters }, "set audio filters", "filter_graph" },
>  #endif
>      { "rdftspeed", OPT_INT | HAS_ARG| OPT_AUDIO | OPT_EXPERT, { &rdftspeed }, "rdft speed", "msecs" },
>      { "showmode", HAS_ARG, { .func_arg = opt_show_mode}, "select show mode (0 = video, 1 = waves, 2 = RDFT)", "mode" },

No more comments from me and I assume it has been tested.

Looking forward for the patch to be applied, many thanks.
-- 
FFmpeg = Funny and Fancy MultiPurpose Ecstatic Gorilla


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