[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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