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author | François Revol <revol@free.fr> | 2007-02-13 18:26:14 +0000 |
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committer | François Revol <revol@free.fr> | 2007-02-13 18:26:14 +0000 |
commit | 8fa36ae09dddb1b639b4df5d505c0dbcf4e916e4 (patch) | |
tree | 551ead2b59bdc4b1855fb60d6c2ce6a2c7787f15 /libavformat/beosaudio.cpp | |
parent | bcdf0d269748e2059ffa789033cd6e41739891fc (diff) | |
download | ffmpeg-8fa36ae09dddb1b639b4df5d505c0dbcf4e916e4.tar.gz |
This fixes error handling for BeOS, removing the need for some ifdefs.
AVERROR_ defines are moved to avcodec.h as they are needed in there as well. Feel free to move that to avutil/common.h.
Bumped up avcodec/format version numbers as though it's binary compatible we will want to rebuild apps as error values changed.
Please from now on use return AVERROR(EFOO) instead of the ugly return -EFOO in your code.
This also removes the need for berrno.h.
Originally committed as revision 7965 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavformat/beosaudio.cpp')
-rw-r--r-- | libavformat/beosaudio.cpp | 22 |
1 files changed, 11 insertions, 11 deletions
diff --git a/libavformat/beosaudio.cpp b/libavformat/beosaudio.cpp index 6ac45ebb29..ae77809747 100644 --- a/libavformat/beosaudio.cpp +++ b/libavformat/beosaudio.cpp @@ -194,15 +194,15 @@ static int audio_open(AudioData *s, int is_output, const char *audio_device) #ifndef HAVE_BSOUNDRECORDER if (!is_output) - return -EIO; /* not for now */ + return AVERROR(EIO); /* not for now */ #endif s->input_sem = create_sem(AUDIO_BUFFER_SIZE, "ffmpeg_ringbuffer_input"); if (s->input_sem < B_OK) - return -EIO; + return AVERROR(EIO); s->output_sem = create_sem(0, "ffmpeg_ringbuffer_output"); if (s->output_sem < B_OK) { delete_sem(s->input_sem); - return -EIO; + return AVERROR(EIO); } s->input_index = 0; s->output_index = 0; @@ -226,7 +226,7 @@ static int audio_open(AudioData *s, int is_output, const char *audio_device) delete_sem(s->input_sem); if (s->output_sem) delete_sem(s->output_sem); - return -EIO; + return AVERROR(EIO); } s->codec_id = (iformat.byte_order == B_MEDIA_LITTLE_ENDIAN)?CODEC_ID_PCM_S16LE:CODEC_ID_PCM_S16BE; s->channels = iformat.channel_count; @@ -252,7 +252,7 @@ static int audio_open(AudioData *s, int is_output, const char *audio_device) delete_sem(s->input_sem); if (s->output_sem) delete_sem(s->output_sem); - return -EIO; + return AVERROR(EIO); } s->player->SetCookie(s); s->player->SetVolume(1.0); @@ -293,7 +293,7 @@ static int audio_write_header(AVFormatContext *s1) s->channels = st->codec->channels; ret = audio_open(s, 1, NULL); if (ret < 0) - return -EIO; + return AVERROR(EIO); return 0; } @@ -315,7 +315,7 @@ lat1 = s->player->Latency(); int amount; len = MIN(size, AUDIO_BLOCK_SIZE); if (acquire_sem_etc(s->input_sem, len, B_CAN_INTERRUPT, 0LL) < B_OK) - return -EIO; + return AVERROR(EIO); amount = MIN(len, (AUDIO_BUFFER_SIZE - s->input_index)); memcpy(&s->buffer[s->input_index], buf, amount); s->input_index += amount; @@ -356,7 +356,7 @@ static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap) st = av_new_stream(s1, 0); if (!st) { - return -ENOMEM; + return AVERROR(ENOMEM); } s->sample_rate = ap->sample_rate; s->channels = ap->channels; @@ -364,7 +364,7 @@ static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap) ret = audio_open(s, 0, ap->device); if (ret < 0) { av_free(st); - return -EIO; + return AVERROR(EIO); } /* take real parameters */ st->codec->codec_type = CODEC_TYPE_AUDIO; @@ -384,7 +384,7 @@ static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt) status_t err; if (av_new_packet(pkt, s->frame_size) < 0) - return -EIO; + return AVERROR(EIO); buf = (unsigned char *)pkt->data; size = pkt->size; while (size > 0) { @@ -393,7 +393,7 @@ static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt) while ((err=acquire_sem_etc(s->output_sem, len, B_CAN_INTERRUPT, 0LL)) == B_INTERRUPTED); if (err < B_OK) { av_free_packet(pkt); - return -EIO; + return AVERROR(EIO); } amount = MIN(len, (AUDIO_BUFFER_SIZE - s->output_index)); memcpy(buf, &s->buffer[s->output_index], amount); |