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| author | reimar <reimar@b3059339-0415-0410-9bf9-f77b7e298cf2> | 2010-07-15 17:59:46 +0000 |
|---|---|---|
| committer | Uoti Urpala <uau@glyph.nonexistent.invalid> | 2010-11-02 04:14:44 +0200 |
| commit | 5ed772b9cddc4c0de6762e223428b3e36eceefff (patch) | |
| tree | 338d1623dd003d777b17c3c64c90049b4c9ebd2c /libmpcodecs/dec_audio.c | |
| parent | 6be6ba40946d333e9d2cf741ba401366e9081c79 (diff) | |
| download | mpv-5ed772b9cddc4c0de6762e223428b3e36eceefff.tar.bz2 mpv-5ed772b9cddc4c0de6762e223428b3e36eceefff.tar.xz | |
audio: support parameter changes (e.g. channel count) during playback
Add support for parameter changes (e.g. channel count) during playback.
This makes decoding AC3 files that switch between 2 and 6 channels
work reasonably well even with -channels 6 and ffac3 decoder.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@31737 b3059339-0415-0410-9bf9-f77b7e298cf2
Fix typo in error message: ACC -> AAC
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32473 b3059339-0415-0410-9bf9-f77b7e298cf2
Avoid printing AAC with SBR warning on every decode call, instead print
it only after every decoder reconfiguration.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32476 b3059339-0415-0410-9bf9-f77b7e298cf2
Diffstat (limited to 'libmpcodecs/dec_audio.c')
| -rw-r--r-- | libmpcodecs/dec_audio.c | 16 |
1 files changed, 12 insertions, 4 deletions
diff --git a/libmpcodecs/dec_audio.c b/libmpcodecs/dec_audio.c index 218203272d..ade2429dc0 100644 --- a/libmpcodecs/dec_audio.c +++ b/libmpcodecs/dec_audio.c @@ -378,13 +378,20 @@ static int filter_n_bytes(sh_audio_t *sh, int len) int error = 0; // Decode more bytes if needed + int old_samplerate = sh->samplerate; + int old_channels = sh->channels; + int old_sample_format = sh->sample_format; while (sh->a_buffer_len < len) { unsigned char *buf = sh->a_buffer + sh->a_buffer_len; int minlen = len - sh->a_buffer_len; int maxlen = sh->a_buffer_size - sh->a_buffer_len; int ret = sh->ad_driver->decode_audio(sh, buf, minlen, maxlen); - if (ret <= 0) { - error = -1; + int format_change = sh->samplerate != old_samplerate + || sh->channels != old_channels + || sh->sample_format != old_sample_format; + if (ret <= 0 || format_change) { + error = format_change ? -2 : -1; + // samples from format-changing call get discarded too len = sh->a_buffer_len; break; } @@ -467,8 +474,9 @@ int decode_audio(sh_audio_t *sh_audio, int minlen) /* if this iteration does not fill buffer, we must have lots * of buffering in filters */ huge_filter_buffer = 1; - if (filter_n_bytes(sh_audio, declen) < 0) - return -1; + int res = filter_n_bytes(sh_audio, declen); + if (res < 0) + return res; } return 0; } |