On 03/28/2011 04:54 PM, Alex Converse wrote:
> ---
> doc/ffmpeg.texi | 2 ++
> ffmpeg.c | 47 +++++++++++++++++++++++++++++++++++++++++++++--
> 2 files changed, 47 insertions(+), 2 deletions(-)
>
>
>
> 0001-Add-an-apad-option-to-ffmpeg-to-pad-audio-to-video-l.patch
>
>
> diff --git a/doc/ffmpeg.texi b/doc/ffmpeg.texi
> index 21c6f2c..2927017 100644
> --- a/doc/ffmpeg.texi
> +++ b/doc/ffmpeg.texi
> @@ -579,6 +579,8 @@ ffmpeg -i file.mpg -vcodec copy -acodec ac3 -ab 384k
> test.mpg -acodec mp2 -ab 19
> @end example
> @item -alang @var{code}
> Set the ISO 639 language code (3 letters) of the current audio stream.
> +@item -apad
> +Pad audio to video length.
> @end table
>
> @section Advanced Audio options:
> diff --git a/ffmpeg.c b/ffmpeg.c
> index 5e50db3..84a278e 100644
> --- a/ffmpeg.c
> +++ b/ffmpeg.c
> @@ -176,6 +176,7 @@ static int64_t channel_layout = 0;
> #define QSCALE_NONE -99999
> static float audio_qscale = QSCALE_NONE;
> static int audio_disable = 0;
> +static int audio_pad = 0;
> static int audio_channels = 1;
> static char *audio_codec_name = NULL;
> static unsigned int audio_codec_tag = 0;
> @@ -297,6 +298,7 @@ typedef struct AVOutputStream {
> int reformat_pair;
> AVAudioConvert *reformat_ctx;
> AVFifoBuffer *fifo; /* for compression: one audio fifo per codec */
> +
> FILE *logfile;
> } AVOutputStream;
>
> @@ -702,7 +704,8 @@ static void write_frame(AVFormatContext *s, AVPacket
> *pkt, AVCodecContext *avctx
> static void do_audio_out(AVFormatContext *s,
> AVOutputStream *ost,
> AVInputStream *ist,
> - unsigned char *buf, int size)
> + unsigned char *buf, int size,
> + int skip_resample_reformat)
> {
> uint8_t *buftmp;
> int64_t audio_out_size, audio_buf_size;
> @@ -739,6 +742,7 @@ need_realloc:
> ffmpeg_exit(1);
> }
>
> + if (!skip_resample_reformat) {
> if (enc->channels != dec->channels)
> ost->audio_resample = 1;
>
> @@ -871,6 +875,10 @@ need_realloc:
> buftmp = audio_buf;
> size_out = len*osize;
> }
> + } else {
> + buftmp = buf;
> + size_out = size;
> + }
>
> /* now encode as many frames as possible */
> if (enc->frame_size > 1) {
> @@ -1603,7 +1611,7 @@ static int output_packet(AVInputStream *ist, int
> ist_index,
> av_assert0(ist->decoding_needed);
> switch(ost->st->codec->codec_type) {
> case AVMEDIA_TYPE_AUDIO:
> - do_audio_out(os, ost, ist, decoded_data_buf,
> decoded_data_size);
> + do_audio_out(os, ost, ist, decoded_data_buf,
> decoded_data_size, 0);
> break;
> case AVMEDIA_TYPE_VIDEO:
> #if CONFIG_AVFILTER
> @@ -1697,8 +1705,18 @@ static int output_packet(AVInputStream *ist, int
> ist_index,
> discard_packet:
> if (pkt == NULL) {
> /* EOF handling */
> + double vpts = 0.0;
> +
> + for(i=0;i<nb_ostreams;i++) {
> + ost = ost_table[i];
> + if(ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
> + double opts = ost->st->pts.val * av_q2d(ost->st->time_base);
> + vpts = FFMAX(opts, vpts);
> + }
> + }
>
> for(i=0;i<nb_ostreams;i++) {
> + int ost_pad_audio = audio_pad;
> ost = ost_table[i];
> if (ost->source_index == ist_index) {
> AVCodecContext *enc= ost->st->codec;
> @@ -1713,13 +1731,37 @@ static int output_packet(AVInputStream *ist, int
> ist_index,
> for(;;) {
> AVPacket pkt;
> int fifo_bytes;
> + double opts;
> av_init_packet(&pkt);
> pkt.stream_index= ost->index;
>
> switch(ost->st->codec->codec_type) {
> case AVMEDIA_TYPE_AUDIO:
> + opts = ost->st->pts.val *
> av_q2d(ost->st->time_base);
> fifo_bytes = av_fifo_size(ost->fifo);
> ret = 0;
> +
> + if (ost_pad_audio && opts < vpts) {
> + int osize =
> av_get_bits_per_sample_fmt(enc->sample_fmt) >> 3;
> + int frame_bytes =
> enc->frame_size*osize*enc->channels;
> + ost_pad_audio = 0;
> + if (samples_size < frame_bytes)
> + ffmpeg_exit(1);
> + memset(samples, 0, frame_bytes);
> + /* finish the current frame in the fifo,
> then send whole frames */
> + if (fifo_bytes > 0) {
> + do_audio_out(os, ost, ist, samples,
> frame_bytes-fifo_bytes, 1);
> + opts = ost->st->pts.val *
> av_q2d(ost->st->time_base);
> + }
> + while (opts < vpts) {
> + do_audio_out(os, ost, ist, samples,
> frame_bytes, 1);
> + opts = ost->st->pts.val *
> av_q2d(ost->st->time_base);
> + }
> + fifo_bytes = av_fifo_size(ost->fifo);
> + if (fifo_bytes != 0)
> + ffmpeg_exit(1);
> + }
> +
> /* encode any samples remaining in fifo */
> if (fifo_bytes > 0) {
> int osize =
> av_get_bits_per_sample_fmt(enc->sample_fmt) >> 3;
> @@ -4233,6 +4275,7 @@ static const OptionDef options[] = {
> { "vol", OPT_INT | HAS_ARG | OPT_AUDIO, {(void*)&audio_volume}, "change
> audio volume (256=normal)" , "volume" }, //
> { "newaudio", OPT_AUDIO | OPT_FUNC2, {(void*)opt_new_stream}, "add a new
> audio stream to the current output stream" },
> { "alang", HAS_ARG | OPT_STRING | OPT_AUDIO, {(void *)&audio_language},
> "set the ISO 639 language code (3 letters) of the current audio stream" ,
> "code" },
> + { "apad", OPT_BOOL | OPT_AUDIO, {(void*)&audio_pad}, "pad audio to video
> length", "pad" },
> { "sample_fmt", HAS_ARG | OPT_EXPERT | OPT_AUDIO,
> {(void*)opt_audio_sample_fmt}, "set sample format, 'list' as argument shows
> all the sample formats supported", "format" },
>
> /* subtitle options */
Based on how you're using skip_resample_reformat, it seems this patch
doesn't work when changing sample format or sample rate when encoding,
correct?
-Justin
_______________________________________________
libav-devel mailing list
[email protected]
https://lists.libav.org/mailman/listinfo/libav-devel
