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

[Timothy Gu]

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.

> +    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
>
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