[FFmpeg-devel] [PATCH] avdevice/alsa: simplify passing ff_alsa_open a channel layout

[James Almer <jamrial@gmail.com>]

This also ensures the layout set during the indev init is used instead of the
blank one in st->codecpar.

Signed-off-by: James Almer <jamrial@gmail.com>
---
 libavdevice/alsa.c     | 14 +++++++-------
 libavdevice/alsa.h     |  4 ++--
 libavdevice/alsa_dec.c |  2 +-
 libavdevice/alsa_enc.c |  2 +-
 4 files changed, 11 insertions(+), 11 deletions(-)

diff --git a/libavdevice/alsa.c b/libavdevice/alsa.c
index d62ccc09c6..cfdb28ff49 100644
--- a/libavdevice/alsa.c
+++ b/libavdevice/alsa.c
@@ -127,7 +127,8 @@ switch(format) {\
     case FORMAT_F32: s->reorder_func = alsa_reorder_f32_out_ ##layout;   break;\
 }
 
-static av_cold int find_reorder_func(AlsaData *s, int codec_id, AVChannelLayout *layout, int out)
+static av_cold int find_reorder_func(AlsaData *s, int codec_id,
+                                     const AVChannelLayout *layout, int out)
 {
     int format;
 
@@ -172,10 +173,9 @@ static av_cold int find_reorder_func(AlsaData *s, int codec_id, AVChannelLayout
 
 av_cold int ff_alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode,
                          unsigned int *sample_rate,
-                         int channels, enum AVCodecID *codec_id)
+                         const AVChannelLayout *layout, enum AVCodecID *codec_id)
 {
     AlsaData *s = ctx->priv_data;
-    AVChannelLayout *layout = &ctx->streams[0]->codecpar->ch_layout;
     const char *audio_device;
     int res, flags = 0;
     snd_pcm_format_t format;
@@ -193,7 +193,7 @@ av_cold int ff_alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode,
         av_log(ctx, AV_LOG_ERROR, "sample format 0x%04x is not supported\n", *codec_id);
         return AVERROR(ENOSYS);
     }
-    s->frame_size = av_get_bits_per_sample(*codec_id) / 8 * channels;
+    s->frame_size = av_get_bits_per_sample(*codec_id) / 8 * layout->nb_channels;
 
     if (ctx->flags & AVFMT_FLAG_NONBLOCK) {
         flags = SND_PCM_NONBLOCK;
@@ -240,10 +240,10 @@ av_cold int ff_alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode,
         goto fail;
     }
 
-    res = snd_pcm_hw_params_set_channels(h, hw_params, channels);
+    res = snd_pcm_hw_params_set_channels(h, hw_params, layout->nb_channels);
     if (res < 0) {
         av_log(ctx, AV_LOG_ERROR, "cannot set channel count to %d (%s)\n",
-               channels, snd_strerror(res));
+               layout->nb_channels, snd_strerror(res));
         goto fail;
     }
 
@@ -277,7 +277,7 @@ av_cold int ff_alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode,
 
     snd_pcm_hw_params_free(hw_params);
 
-    if (channels > 2 && layout->order != AV_CHANNEL_ORDER_UNSPEC) {
+    if (layout->nb_channels > 2 && layout->order != AV_CHANNEL_ORDER_UNSPEC) {
         if (find_reorder_func(s, *codec_id, layout, mode == SND_PCM_STREAM_PLAYBACK) < 0) {
             char name[128];
             av_channel_layout_describe(layout, name, sizeof(name));
diff --git a/libavdevice/alsa.h b/libavdevice/alsa.h
index 3e1ba31384..d3dfa478c5 100644
--- a/libavdevice/alsa.h
+++ b/libavdevice/alsa.h
@@ -72,7 +72,7 @@ typedef struct AlsaData {
  * @param mode either SND_PCM_STREAM_CAPTURE or SND_PCM_STREAM_PLAYBACK
  * @param sample_rate in: requested sample rate;
  *                    out: actually selected sample rate
- * @param channels number of channels
+ * @param layout channel layout
  * @param codec_id in: requested AVCodecID or AV_CODEC_ID_NONE;
  *                 out: actually selected AVCodecID, changed only if
  *                 AV_CODEC_ID_NONE was requested
@@ -82,7 +82,7 @@ typedef struct AlsaData {
 av_warn_unused_result
 int ff_alsa_open(AVFormatContext *s, snd_pcm_stream_t mode,
                  unsigned int *sample_rate,
-                 int channels, enum AVCodecID *codec_id);
+                 const AVChannelLayout *layout, enum AVCodecID *codec_id);
 
 /**
  * Close the ALSA PCM.
diff --git a/libavdevice/alsa_dec.c b/libavdevice/alsa_dec.c
index f0738e3dea..63409a7785 100644
--- a/libavdevice/alsa_dec.c
+++ b/libavdevice/alsa_dec.c
@@ -80,7 +80,7 @@ static av_cold int audio_read_header(AVFormatContext *s1)
     }
 #endif
 
-    ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->ch_layout.nb_channels,
+    ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, &s->ch_layout,
         &codec_id);
     if (ret < 0) {
         return AVERROR(EIO);
diff --git a/libavdevice/alsa_enc.c b/libavdevice/alsa_enc.c
index 0b4c7834f7..971cff688c 100644
--- a/libavdevice/alsa_enc.c
+++ b/libavdevice/alsa_enc.c
@@ -66,7 +66,7 @@ static av_cold int audio_write_header(AVFormatContext *s1)
     sample_rate = st->codecpar->sample_rate;
     codec_id    = st->codecpar->codec_id;
     res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate,
-        st->codecpar->ch_layout.nb_channels, &codec_id);
+        &st->codecpar->ch_layout, &codec_id);
     if (sample_rate != st->codecpar->sample_rate) {
         av_log(s1, AV_LOG_ERROR,
                "sample rate %d not available, nearest is %d\n",
-- 
2.48.1

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