[FFmpeg-devel] [PATCH 2/2] lavd/alsa: add stream validation

[Timothy Gu]

On Oct 25, 2013 5:39 PM, "Timothy Gu" <timothygu99 at gmail.com> wrote:
>
>
> On Oct 25, 2013 4:40 PM, "Lukasz Marek" <lukasz.m.luki at gmail.com> wrote:
> >
> > Don't trust provided streams. Find first audio stream and use it.
> > Make a warning if more than one.
> >
> > Signed-off-by: Lukasz Marek <lukasz.m.luki at gmail.com>
> > ---
> >  libavdevice/alsa-audio-enc.c |   20 +++++++++++++++++---
> >  libavdevice/alsa-audio.h     |    1 +
> >  2 files changed, 18 insertions(+), 3 deletions(-)
> >
> > diff --git a/libavdevice/alsa-audio-enc.c b/libavdevice/alsa-audio-enc.c
> > index 4d0e17b..1d57462 100644
> > --- a/libavdevice/alsa-audio-enc.c
> > +++ b/libavdevice/alsa-audio-enc.c
> > @@ -51,8 +51,22 @@ static av_cold int
audio_write_header(AVFormatContext *s1)
> >      unsigned int sample_rate;
> >      enum AVCodecID codec_id;
> >      int res;
> > +    unsigned int i;
> > +
> > +    for (i = 0; i < s1->nb_streams; i++) {
> > +        if (s1->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
> > +            st = s1->streams[i];
> > +            s->index = i;
> > +        } else {
> > +            av_log(s1, AV_LOG_WARNING, "More than one audio stream
found. First one is used.\n");
> > +            break;
> > +        }
> > +    }
>
> The logic is wrong here. So if stream[0]->codec->codec_type !=
AVMEDIA_TYPE_AUDIO,
> it will report that "More than one audio stream is found".
> And if stream[1] is audio too, it will overwrite s->index to be 2.

I mean overwrite index to be 1.

>
> > +    if (!st) {
> > +        av_log(s, AV_LOG_ERROR, "No audio stream found.\n");
> > +        return AVERROR(EINVAL);
> > +    }
> >
> > -    st = s1->streams[0];
> >      sample_rate = st->codec->sample_rate;
> >      codec_id    = st->codec->codec_id;
> >      res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate,
> > @@ -80,7 +94,7 @@ static int audio_write_packet(AVFormatContext *s1,
AVPacket *pkt)
> >      uint8_t *buf = pkt->data;
> >
> >      if (!(s->timestamp_diff = pkt->duration)) {
> > -        AVStream *st = s1->streams[0];
> > +        AVStream *st = s1->streams[s->index];
> >          AVCodecContext *codec_ctx = st->codec;
> >          /*XXX: no need to recalculate: 1/sample_rate ==
avprinv_set_pts_info() */
> >          s->timestamp_diff = pkt->size /
(av_get_bytes_per_sample(codec_ctx->sample_fmt) * codec_ctx->channels);
> > @@ -118,7 +132,7 @@ static void
audio_get_output_timestamp(AVFormatContext *s1, int stream,
> >      snd_pcm_sframes_t delay = 0;
> >      *wall = av_gettime();
> >      snd_pcm_delay(s->h, &delay);
> > -    *dts = s1->streams[0]->cur_dts + s->timestamp_diff - delay;
> > +    *dts = s1->streams[s->index]->cur_dts + s->timestamp_diff - delay;
> >  }
> >
> >  AVOutputFormat ff_alsa_muxer = {
> > diff --git a/libavdevice/alsa-audio.h b/libavdevice/alsa-audio.h
> > index b9670b9..1173d2b 100644
> > --- a/libavdevice/alsa-audio.h
> > +++ b/libavdevice/alsa-audio.h
> > @@ -58,6 +58,7 @@ typedef struct AlsaData {
> >      void *reorder_buf;
> >      int reorder_buf_size; ///< in frames
> >      int64_t timestamp_diff; ///< duration of last packet, need to
calculate timestamp
> > +    unsigned int index;     ///< index of the firstfound audio
> >  } AlsaData;
> >
> >  /**
> > --
> > 1.7.10.4
> >
> > _______________________________________________
> > ffmpeg-devel mailing list
> > ffmpeg-devel at ffmpeg.org
> > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel