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

[Lukasz Marek]

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;
+        }
+    }
+    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