[FFmpeg-devel] [PATCHv3] ffplay: add -af option
Marton Balint
cus at passwd.hu
Sat Mar 16 00:40:25 CET 2013
On Fri, 15 Mar 2013, Stefano Sabatini wrote:
> 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
>
Added a slightly shorter comment which covers this.
>
>> + 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
Fixed.
>
>> + 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
Fixed.
>
>> + 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
Fixed.
>
>> + 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?
No, the way I see it it's not, so I've removed it. However, probably some
code factorization could be done because the code in
libavutil/frame.c:get_frame_defaults and
libavcodec/utils.c:avcoded_get_frame_defaults are almost the same, and by
factoring out the common part we could ensure that these two will not
diverge in the future.
>
>> + 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.
I've sent a seperate email with the updated patch. I will post a merge
request in a day or two. Thanks to you too for the initial patch!
Regards,
Marton
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