[FFmpeg-cvslog] ALSA: implement channel layout for playback.
Nicolas George
git at videolan.org
Sat Mar 26 17:00:28 CET 2011
ffmpeg | branch: master | Nicolas George <nicolas.george at normalesup.org> | Thu Mar 10 12:10:34 2011 +0100| [813dbb44424846274a318bfc6d02a53b4189f884] | committer: Nicolas George
ALSA: implement channel layout for playback.
Currently, only S16 quad, 5.1 and 7.1 are implemented.
Implementing support for other formats/layouts and capture should be
straightforward.
7.1 support by Carl Eugen Hoyos.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=813dbb44424846274a318bfc6d02a53b4189f884
---
libavdevice/alsa-audio-common.c | 90 +++++++++++++++++++++++++++++++++++++++
libavdevice/alsa-audio-enc.c | 10 ++++-
libavdevice/alsa-audio.h | 7 +++
3 files changed, 106 insertions(+), 1 deletions(-)
diff --git a/libavdevice/alsa-audio-common.c b/libavdevice/alsa-audio-common.c
index 38cb0de..4edd25b 100644
--- a/libavdevice/alsa-audio-common.c
+++ b/libavdevice/alsa-audio-common.c
@@ -43,6 +43,59 @@ static av_cold snd_pcm_format_t codec_id_to_pcm_format(int codec_id)
}
}
+static void alsa_reorder_s16_out_51(const void *in_v, void *out_v, int n)
+{
+ const int16_t *in = in_v;
+ int16_t *out = out_v;
+
+ while (n-- > 0) {
+ out[0] = in[0];
+ out[1] = in[1];
+ out[2] = in[4];
+ out[3] = in[5];
+ out[4] = in[2];
+ out[5] = in[3];
+ in += 6;
+ out += 6;
+ }
+}
+
+static void alsa_reorder_s16_out_71(const void *in_v, void *out_v, int n)
+{
+ const int16_t *in = in_v;
+ int16_t *out = out_v;
+
+ while (n-- > 0) {
+ out[0] = in[0];
+ out[1] = in[1];
+ out[2] = in[4];
+ out[3] = in[5];
+ out[4] = in[2];
+ out[5] = in[3];
+ out[6] = in[6];
+ out[7] = in[7];
+ in += 8;
+ out += 8;
+ }
+}
+
+#define REORDER_DUMMY ((void *)1)
+
+static av_cold ff_reorder_func find_reorder_func(int codec_id,
+ int64_t layout,
+ int out)
+{
+ return
+ codec_id == CODEC_ID_PCM_S16LE || codec_id == CODEC_ID_PCM_S16BE ?
+ layout == AV_CH_LAYOUT_QUAD ? REORDER_DUMMY :
+ layout == AV_CH_LAYOUT_5POINT1_BACK ?
+ out ? alsa_reorder_s16_out_51 : NULL :
+ layout == AV_CH_LAYOUT_7POINT1 ?
+ out ? alsa_reorder_s16_out_71 : NULL :
+ NULL :
+ NULL;
+}
+
av_cold int ff_alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode,
unsigned int *sample_rate,
int channels, enum CodecID *codec_id)
@@ -54,6 +107,7 @@ av_cold int ff_alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode,
snd_pcm_t *h;
snd_pcm_hw_params_t *hw_params;
snd_pcm_uframes_t buffer_size, period_size;
+ int64_t layout = ctx->streams[0]->codec->channel_layout;
if (ctx->filename[0] == 0) audio_device = "default";
else audio_device = ctx->filename;
@@ -146,6 +200,26 @@ av_cold int ff_alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode,
snd_pcm_hw_params_free(hw_params);
+ if (channels > 2 && layout) {
+ s->reorder_func = find_reorder_func(*codec_id, layout,
+ mode == SND_PCM_STREAM_PLAYBACK);
+ if (s->reorder_func == REORDER_DUMMY) {
+ s->reorder_func = NULL;
+ } else if (s->reorder_func) {
+ s->reorder_buf_size = buffer_size;
+ s->reorder_buf = av_malloc(s->reorder_buf_size * s->frame_size);
+ if (!s->reorder_buf)
+ goto fail1;
+ } else {
+ char name[16];
+ av_get_channel_layout_string(name, sizeof(name), channels, layout);
+ av_log(ctx, AV_LOG_WARNING,
+ "ALSA channel layout unknown or unimplemented for %s %s.\n",
+ name,
+ mode == SND_PCM_STREAM_PLAYBACK ? "playback" : "capture");
+ }
+ }
+
s->h = h;
return 0;
@@ -160,6 +234,7 @@ av_cold int ff_alsa_close(AVFormatContext *s1)
{
AlsaData *s = s1->priv_data;
+ av_freep(s->reorder_buf);
snd_pcm_close(s->h);
return 0;
}
@@ -184,3 +259,18 @@ int ff_alsa_xrun_recover(AVFormatContext *s1, int err)
}
return err;
}
+
+int ff_alsa_extend_reorder_buf(AlsaData *s, int min_size)
+{
+ int size = s->reorder_buf_size;
+ void *r;
+
+ while (size < min_size)
+ size *= 2;
+ r = av_realloc(s->reorder_buf, size * s->frame_size);
+ if (!r)
+ return AVERROR(ENOMEM);
+ s->reorder_buf = r;
+ s->reorder_buf_size = size;
+ return 0;
+}
diff --git a/libavdevice/alsa-audio-enc.c b/libavdevice/alsa-audio-enc.c
index 83a179e..0bc53b6 100644
--- a/libavdevice/alsa-audio-enc.c
+++ b/libavdevice/alsa-audio-enc.c
@@ -76,7 +76,15 @@ static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
int size = pkt->size;
uint8_t *buf = pkt->data;
- while((res = snd_pcm_writei(s->h, buf, size / s->frame_size)) < 0) {
+ size /= s->frame_size;
+ if (s->reorder_func) {
+ if (size > s->reorder_buf_size)
+ if (ff_alsa_extend_reorder_buf(s, size))
+ return AVERROR(ENOMEM);
+ s->reorder_func(buf, s->reorder_buf, size);
+ buf = s->reorder_buf;
+ }
+ while ((res = snd_pcm_writei(s->h, buf, size)) < 0) {
if (res == -EAGAIN) {
return AVERROR(EAGAIN);
diff --git a/libavdevice/alsa-audio.h b/libavdevice/alsa-audio.h
index 9493831..8c7c516 100644
--- a/libavdevice/alsa-audio.h
+++ b/libavdevice/alsa-audio.h
@@ -39,10 +39,15 @@
other formats */
#define DEFAULT_CODEC_ID AV_NE(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE)
+typedef void (*ff_reorder_func)(const void *, void *, int);
+
typedef struct {
snd_pcm_t *h;
int frame_size; ///< preferred size for reads and writes
int period_size; ///< bytes per sample * channels
+ ff_reorder_func reorder_func;
+ void *reorder_buf;
+ int reorder_buf_size; ///< in frames
} AlsaData;
/**
@@ -82,4 +87,6 @@ int ff_alsa_close(AVFormatContext *s1);
*/
int ff_alsa_xrun_recover(AVFormatContext *s1, int err);
+int ff_alsa_extend_reorder_buf(AlsaData *s, int size);
+
#endif /* AVDEVICE_ALSA_AUDIO_H */
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