[FFmpeg-devel] [PATCH 3/3] ffplay: add -af option
Marton Balint
cus at passwd.hu
Sun Mar 3 02:28:38 CET 2013
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 | 171 ++++++++++++++++++++++++++++++++++++++++++++++++++++++-
2 files changed, 175 insertions(+), 2 deletions(-)
diff --git a/doc/ffplay.texi b/doc/ffplay.texi
index 8d6abee..95eee3c 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 3343ed3..074ba6e 100644
--- a/ffplay.c
+++ b/ffplay.c
@@ -128,6 +128,7 @@ typedef struct VideoPicture {
#if CONFIG_AVFILTER
AVFilterBufferRef *picref;
+ AVFilterBufferRef *samplesref;
#endif
} VideoPicture;
@@ -194,7 +195,11 @@ 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;
@@ -258,6 +263,10 @@ typedef struct VideoState {
AVFilterContext *out_video_filter; // the last filter in the video chain
int use_dr1;
FrameBuffer *buffer_pool;
+
+ 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;
@@ -314,6 +323,7 @@ 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 */
@@ -327,6 +337,24 @@ 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)
+{
+ 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) {
+ 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)
@@ -1798,6 +1826,70 @@ 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;
+ 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;
+ return 0;
+
+fail:
+ avfilter_graph_free(&is->agraph);
+ return ret;
+}
#endif /* CONFIG_AVFILTER */
static int video_thread(void *arg)
@@ -2096,6 +2188,7 @@ 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 */
@@ -2136,6 +2229,52 @@ 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
+ {
+ AVFilterBufferRef *samplesref;
+ 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)) < 0)
+ return ret;
+ if ((ret = av_buffersink_get_buffer_ref(is->out_audio_filter,
+ &samplesref, 0)) < 0)
+ return ret;
+ if ((ret = avfilter_copy_buf_props(is->frame, samplesref)) < 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,
@@ -2202,7 +2341,7 @@ 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
@@ -2346,6 +2485,8 @@ 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)
@@ -2399,8 +2540,29 @@ 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;
@@ -2472,6 +2634,9 @@ 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);
@@ -2864,6 +3029,7 @@ 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) {
@@ -3272,6 +3438,7 @@ 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" },
--
1.7.10.4
More information about the ffmpeg-devel
mailing list