[FFmpeg-devel] [PATCH+RFC] AVFrame for audio
Michael Niedermayer
michaelni
Wed Nov 24 19:00:12 CET 2010
On Sat, Nov 13, 2010 at 01:13:09PM -0500, Justin Ruggles wrote:
> Michael Niedermayer wrote:
>
> > On Mon, Nov 01, 2010 at 05:19:00PM -0400, Justin Ruggles wrote:
> >> Michael Niedermayer wrote:
> >>
> >>> On Sat, Oct 30, 2010 at 08:06:42AM -0400, Justin Ruggles wrote:
> >>>> Michael Niedermayer wrote:
> >>>>
> >>>>> On Thu, Oct 28, 2010 at 06:58:11PM -0400, Justin Ruggles wrote:
> >>>>>> Michael Niedermayer wrote:
> >>>>>>
> >>>>>>> On Wed, Oct 27, 2010 at 10:13:10PM -0400, Justin Ruggles wrote:
> >>>>>>>> Michael Niedermayer wrote:
> >>>>>>>>
> >>>>>>>>> On Tue, Oct 26, 2010 at 09:31:13PM -0400, Justin Ruggles wrote:
> >>>>>>>>>> Michael Niedermayer wrote:
> >>>>>>>>>>
> >>>>>>>>>>> On Sun, Oct 17, 2010 at 05:22:54PM -0400, Justin Ruggles wrote:
> >>>>>>>>>>>> Michael Niedermayer wrote:
> >>>>>>>>>>>>
> >>>>>>>>>>>>> On Sat, Oct 16, 2010 at 04:12:26PM -0400, Justin Ruggles wrote:
> >>>>>>>>>>>>>> Michael Niedermayer wrote:
> >>>>>>>>>>>>>>
> >>>>>>>>>>>>>>> On Fri, Oct 15, 2010 at 06:35:01PM -0400, Justin Ruggles wrote:
> >>>>>>>>>>>>>>>> Justin Ruggles wrote:
> >>>>>>>>>>>>>>>>
> >>>>>>>>>>>>>>>>> Michael Niedermayer wrote:
> >>>>>>>>>>>>>>>>>
> >>>>>>>>>>>>>>>>>> On Wed, Oct 13, 2010 at 07:52:12PM -0400, Justin Ruggles wrote:
> >>>>>>>>>>>>>>>>>>> Michael Niedermayer wrote:
> >>>>>>>>>>>>>>>>>>>
> >>>>>>>>>>>>>>>>>>>> On Wed, Oct 06, 2010 at 11:05:26AM -0400, Justin Ruggles wrote:
> >>>>>>>>>>>>>>>>>>>>> Michael Niedermayer wrote:
> >>>>>>>>>>>>>>>>>>>>>
> >>>>>>>>>>>>>>>>>>>>>> On Tue, Oct 05, 2010 at 04:55:12PM -0400, Justin Ruggles wrote:
> >>>>>>>>>>>>>>>>>>>>>>> Michael Niedermayer wrote:
> >>>>>>>>>>>>>>>>>>>>>>>
> >>>>>>>>>>>>>>>>>>>>>>>> On Wed, Sep 29, 2010 at 09:20:04PM -0400, Justin Ruggles wrote:
> >>>>>>>>>>>>>>>>>>>>>>>>> Hi,
> >>>>>>>>>>>>>>>>>>>>>>>>>
> >>>>>>>>>>>>>>>>>>>>>>>>> Peter Ross wrote:
> >>>>>>>>>>>>>>>>>>>>>>>>>
> >>>>>>>>>>>>>>>>>>>>>>>>>> On Thu, Sep 02, 2010 at 07:11:37PM -0400, Justin Ruggles wrote:
> >>>>>>>>>>>>>>>>>>>>>>>>>>> Hi,
> >>>>>>>>>>>>>>>>>>>> [...]
> >>>>>>>>>>>>>>>>>>>>>>>> [...]
> >>>>>>>>>>>>>>>>>>>>>>>>> @@ -644,29 +677,49 @@
> >>>>>>>>>>>>>>>>>>>>>>>>> }
> >>>>>>>>>>>>>>>>>>>>>>>>> #endif
> >>>>>>>>>>>>>>>>>>>>>>>>>
> >>>>>>>>>>>>>>>>>>>>>>>>> +#if LIBAVCODEC_VERSION_MAJOR < 53
> >>>>>>>>>>>>>>>>>>>>>>>>> int attribute_align_arg avcodec_decode_audio3(AVCodecContext *avctx, int16_t *samples,
> >>>>>>>>>>>>>>>>>>>>>>>>> int *frame_size_ptr,
> >>>>>>>>>>>>>>>>>>>>>>>>> AVPacket *avpkt)
> >>>>>>>>>>>>>>>>>>>>>>>>> {
> >>>>>>>>>>>>>>>>>>>>>>>>> + AVFrame frame;
> >>>>>>>>>>>>>>>>>>>>>>>>> + int ret, got_frame = 0;
> >>>>>>>>>>>>>>>>>>>>>>>>> +
> >>>>>>>>>>>>>>>>>>>>>>>>> + avcodec_get_frame_defaults(&frame);
> >>>>>>>>>>>>>>>>>>>>>>>>> +
> >>>>>>>>>>>>>>>>>>>>>>>>> + ret = avcodec_decode_audio4(avctx, &frame, &got_frame, avpkt);
> >>>>>>>>>>>>>>>>>>>>>>>>> +
> >>>>>>>>>>>>>>>>>>>>>>>>> + if (ret >= 0 && got_frame) {
> >>>>>>>>>>>>>>>>>>>>>>>>> + *frame_size_ptr = frame.nb_samples * avctx->channels *
> >>>>>>>>>>>>>>>>>>>>>>>>> + (av_get_bits_per_sample_format(avctx->sample_fmt) / 8);
> >>>>>>>>>>>>>>>>>>>>>>>>> +
> >>>>>>>>>>>>>>>>>>>>>>>>> + /* ensure data will fit in the output buffer */
> >>>>>>>>>>>>>>>>>>>>>>>>> + if (*frame_size_ptr > AVCODEC_MAX_AUDIO_FRAME_SIZE) {
> >>>>>>>>>>>>>>>>>>>>>>>>> + av_log(avctx, AV_LOG_WARNING, "avcodec_decode_audio3 samples "
> >>>>>>>>>>>>>>>>>>>>>>>>> + "truncated to AVCODEC_MAX_AUDIO_FRAME_SIZE\n");
> >>>>>>>>>>>>>>>>>>>>>>>>> + *frame_size_ptr = AVCODEC_MAX_AUDIO_FRAME_SIZE;
> >>>>>>>>>>>>>>>>>>>>>>>>> + }
> >>>>>>>>>>>>>>>>>>>>>>>>> +
> >>>>>>>>>>>>>>>>>>>>>>>>> + memcpy(samples, frame.data[0], *frame_size_ptr);
> >>>>>>>>>>>>>>>>>>>>>>>> the default get_buffer() should return the appropriate
> >>>>>>>>>>>>>>>>>>>>>>>> buffer for this case.
> >>>>>>>>>>>>>>>>>>>>>>> I'm sorry, I don't understand your comment.
> >>>>>>>>>>>>>>>>>>>>>> i mean (non functional psseudocode below to explain the idea)
> >>>>>>>>>>>>>>>>>>>>>> avcodec_decode_audio3(){
> >>>>>>>>>>>>>>>>>>>>>> avctx->foobar= samples;
> >>>>>>>>>>>>>>>>>>>>>> ret = avcodec_decode_audio4(avctx, &frame, &got_frame, avpkt);
> >>>>>>>>>>>>>>>>>>>>>> ...
> >>>>>>>>>>>>>>>>>>>>>> assert(samples == frame.data[0]);
> >>>>>>>>>>>>>>>>>>>>>>
> >>>>>>>>>>>>>>>>>>>>>> -----
> >>>>>>>>>>>>>>>>>>>>>> default_get_buffer(){
> >>>>>>>>>>>>>>>>>>>>>> if(avctx->foobar)
> >>>>>>>>>>>>>>>>>>>>>> return avctx->foobar;
> >>>>>>>>>>>>>>>>>>>>>>
> >>>>>>>>>>>>>>>>>>>>>>
> >>>>>>>>>>>>>>>>>>>>>> (and yes this cannot work for all theoretical decoders)
> >>>>>>>>>>>>>>>>>>>>> I think I get it. So avctx->foobar would be an optional user-supplied
> >>>>>>>>>>>>>>>>>>>>> buffer (avctx->user_buffer?) that default_get_buffer() would return if
> >>>>>>>>>>>>>>>>>>>>> it is non-NULL, right?
> >>>>>>>>>>>>>>>>>>>> yes
> >>>>>>>>>>>>>>>>>>>>
> >>>>>>>>>>>>>>>>>>>>
> >>>>>>>>>>>>>>>>>>>>> The problem I see is this:
> >>>>>>>>>>>>>>>>>>>>> avcodec_decode_audio3() would use avcodec_decode_audio4().
> >>>>>>>>>>>>>>>>>>>>> avcodec_decode_audio4() allocates as large a buffer as is needed through
> >>>>>>>>>>>>>>>>>>>>> get_buffer(), but the avcodec_decode_audio3() API only requires the
> >>>>>>>>>>>>>>>>>>>>> user-supplied buffer to be AVCODEC_MAX_AUDIO_FRAME_SIZE. Couldn't this
> >>>>>>>>>>>>>>>>>>>>> lead to the decoder writing past the end of a user-supplied buffer if it
> >>>>>>>>>>>>>>>>>>>>> isn't large enough? I guess we could also add a field
> >>>>>>>>>>>>>>>>>>>>> avctx->user_buffer_size?
> >>>>>>>>>>>>>>>>>>>> yes, of course
> >>>>>>>>>>>>>>>>>>>> it was just a rough idea ...
> >>>>>>>>>>>>>>>>>>> I'm running into some questions trying to implement the rough idea. The
> >>>>>>>>>>>>>>>>>>> only way I can see this working smoothly is if avcodec_decode_audio3()
> >>>>>>>>>>>>>>>>>>> always sets get/release_buffer to default. Also, either all audio
> >>>>>>>>>>>>>>>>>>> decoders will have to support CODEC_CAP_DR1 (i.e. they always use
> >>>>>>>>>>>>>>>>>>> get/release_buffer) or there needs to be a fallback that will memcpy
> >>>>>>>>>>>>>>>>>>> into the user buffer if CODEC_CAP_DR1 is not supported.
> >>>>>>>>>>>>>>>>>> old API decoders surely dont need to copy with old API.
> >>>>>>>>>>>>>>>>>> old API decoders surely dont need to copy with new API if the api can provide
> >>>>>>>>>>>>>>>>>> a buffer to the decoder (this can be through function argument like its done
> >>>>>>>>>>>>>>>>>> currently)
> >>>>>>>>>>>>>>>>>> new API decoders surely dont need to copy with new API because otherwise the
> >>>>>>>>>>>>>>>>>> API sucks and needs more work
> >>>>>>>>>>>>>>>>>> whats left is new API decoders and used with old API and for this get_buffer()
> >>>>>>>>>>>>>>>>>> should return the user supplied buffer if its large enough and fail if its not
> >>>>>>>>>>>>>>>>>> large enough.
> >>>>>>>>>>>>>>>>>> The case where the user overrides get_buffer() and supplies a user specified
> >>>>>>>>>>>>>>>>>> buffer which its own code doesnt use is a case that id consider user error.
> >>>>>>>>>>>>>>>>> I think I might have been misinterpreting the API. For video decoders,
> >>>>>>>>>>>>>>>>> what does it mean as far as buffer allocation when CODEC_CAP_DR1 is not set?
> >>>>>>>>>>>>>>>> So I think I have this worked out and I don't see how we can avoid a
> >>>>>>>>>>>>>>>> memcpy with the old API when CODEC_CAP_DR1 is not set. There would be
> >>>>>>>>>>>>>>>> no other way to get the data into the correct output buffer.
> >>>>>>>>>>>>>>>>
> >>>>>>>>>>>>>>>> other questions:
> >>>>>>>>>>>>>>>>
> >>>>>>>>>>>>>>>> 1. Should AVCodecContext.user_buffer be supported for video decoders?
> >>>>>>>>>>>>>>> possible but this is seperate, lets not entangle this patch with too many
> >>>>>>>>>>>>>>> other changes
> >>>>>>>>>>>>>>>
> >>>>>>>>>>>>>>>
> >>>>>>>>>>>>>>>> If so, should it be user_buffer[4] and user_buffer_size[4]?
> >>>>>>>>>>>>>>> possible
> >>>>>>>>>>>>>>>
> >>>>>>>>>>>>>>>
> >>>>>>>>>>>>>>>
> >>>>>>>>>>>>>>>> 2. avcodec_default_get_buffer() supports allocating multiple internal
> >>>>>>>>>>>>>>>> buffers. How should that be handled if the buffer is supplied by the
> >>>>>>>>>>>>>>>> user? Don't support multiple buffers? Use the user-supplied buffer
> >>>>>>>>>>>>>>>> just for the first one?
> >>>>>>>>>>>>>>> there are buffer hints (grep for hint in avcodec.h) that specify if a buffer
> >>>>>>>>>>>>>>> will be reused/read/preserved/blah
> >>>>>>>>>>>>>>> the user supplied buffer is likely just valid for this call and cannot be used
> >>>>>>>>>>>>>>> for some cases of the hints. For what remains using the buffer on the first
> >>>>>>>>>>>>>>> call only seems ok
> >>>>>>>>>>>>>> I think I've implemented it in a way that will work even when the
> >>>>>>>>>>>>>> various buffer hints are set. This implementation will not use memcpy
> >>>>>>>>>>>>>> in avcodec_decode_audio3() in the most common case of the decoder
> >>>>>>>>>>>>>> supporting CODEC_CAP_DR1, only needing 1 buffer, and not needing a
> >>>>>>>>>>>>>> buffer larger than AVCODEC_MAX_AUDIO_FRAME_SIZE.
> >>>>>>>>>>>>>>
> >>>>>>>>>>>>>> One thing I'm unsure of is whether to truncate output if it is too large
> >>>>>>>>>>>>>> for avcodec_decode_audio3() (which is done in this patch) or to return
> >>>>>>>>>>>>>> an error instead.
> >>>>>>>>>>>>> I think its better to tell the user straight through an error that there is a
> >>>>>>>>>>>>> problem instead of generating output that contains randomly truncated packets
> >>>>>>>>>>>> Ok. New patch.
> >>>>>>>>>>>>
> >>>>>>>>>>>> -Justin
> >>>>>>>>>>> [...]
> >>>>>>>>>>>> int avcodec_default_get_buffer(AVCodecContext *s, AVFrame *pic){
> >>>>>>>>>>>> int i;
> >>>>>>>>>>>> int w= s->width;
> >>>>>>>>>>>> int h= s->height;
> >>>>>>>>>>>> + int is_video = (s->codec_type == AVMEDIA_TYPE_VIDEO);
> >>>>>>>>>>>> InternalBuffer *buf;
> >>>>>>>>>>>> int *picture_number;
> >>>>>>>>>>>>
> >>>>>>>>>>>> @@ -235,7 +258,8 @@ int avcodec_default_get_buffer(AVCodecContext *s, AVFrame *pic){
> >>>>>>>>>>>> return -1;
> >>>>>>>>>>>> }
> >>>>>>>>>>>>
> >>>>>>>>>>>> - if(av_image_check_size(w, h, 0, s))
> >>>>>>>>>>>> + if(( is_video && av_image_check_size(w, h, 0, s)) ||
> >>>>>>>>>>>> + (!is_video && audio_check_size(s->channels, pic->nb_samples, s->sample_fmt)))
> >>>>>>>>>>>> return -1;
> >>>>>>>>>>>>
> >>>>>>>>>>>> if(s->internal_buffer==NULL){
> >>>>>>>>>>>> @@ -253,17 +277,30 @@ int avcodec_default_get_buffer(AVCodecContext *s, AVFrame *pic){
> >>>>>>>>>>>> picture_number= &(((InternalBuffer*)s->internal_buffer)[INTERNAL_BUFFER_SIZE]).last_pic_num; //FIXME ugly hack
> >>>>>>>>>>>> (*picture_number)++;
> >>>>>>>>>>>>
> >>>>>>>>>>>> - if(buf->base[0] && (buf->width != w || buf->height != h || buf->pix_fmt != s->pix_fmt)){
> >>>>>>>>>>>> + if (buf->base[0]) {
> >>>>>>>>>>>> + if (is_video && (buf->width != w || buf->height != h || buf->pix_fmt != s->pix_fmt)){
> >>>>>>>>>>>> for(i=0; i<4; i++){
> >>>>>>>>>>>> av_freep(&buf->base[i]);
> >>>>>>>>>>>> buf->data[i]= NULL;
> >>>>>>>>>>>> }
> >>>>>>>>>>>> + } else if (!is_video && (buf->channels != s->channels ||
> >>>>>>>>>>>> + buf->nb_samples != pic->nb_samples ||
> >>>>>>>>>>>> + buf->sample_fmt != s->sample_fmt)) {
> >>>>>>>>>>>> + if (buf->base[0] == s->user_buffer) {
> >>>>>>>>>>>> + s->user_buffer_in_use = 0;
> >>>>>>>>>>>> + buf->base[0] = NULL;
> >>>>>>>>>>>> + } else {
> >>>>>>>>>>>> + av_freep(&buf->base[0]);
> >>>>>>>>>>>> + }
> >>>>>>>>>>>> + buf->data[0] = NULL;
> >>>>>>>>>>>> + }
> >>>>>>>>>>>> }
> >>>>>>>>>>>>
> >>>>>>>>>>>> if(buf->base[0]){
> >>>>>>>>>>>> pic->age= *picture_number - buf->last_pic_num;
> >>>>>>>>>>>> buf->last_pic_num= *picture_number;
> >>>>>>>>>>>> }else{
> >>>>>>>>>>>> + if (is_video) {
> >>>>>>>>>>>> int h_chroma_shift, v_chroma_shift;
> >>>>>>>>>>>> int size[4] = {0};
> >>>>>>>>>>>> int tmpsize;
> >>>>>>>>>>>> @@ -327,6 +364,28 @@ int avcodec_default_get_buffer(AVCodecContext *s, AVFrame *pic){
> >>>>>>>>>>>> buf->height = s->height;
> >>>>>>>>>>>> buf->pix_fmt= s->pix_fmt;
> >>>>>>>>>>>> pic->age= 256*256*256*64;
> >>>>>>>>>>>> + } else { /* audio */
> >>>>>>>>>>>> + int buf_size;
> >>>>>>>>>>>> +
> >>>>>>>>>>>> + buf->last_pic_num = -256*256*256*64;
> >>>>>>>>>>>> +
> >>>>>>>>>>>> + buf_size = pic->nb_samples * s->channels *
> >>>>>>>>>>>> + (av_get_bits_per_sample_format(s->sample_fmt) / 8);
> >>>>>>>>>>>> +
> >>>>>>>>>>>> + if (s->user_buffer && !s->user_buffer_in_use && s->user_buffer_size >= buf_size) {
> >>>>>>>>>>>> + buf->base[0] = s->user_buffer;
> >>>>>>>>>>>> + s->user_buffer_in_use = 1;
> >>>>>>>>>>>> + } else {
> >>>>>>>>>>>> + buf->base[0] = av_mallocz(buf_size);
> >>>>>>>>>>>> + if (!buf->base[0])
> >>>>>>>>>>>> + return AVERROR(ENOMEM);
> >>>>>>>>>>>> + }
> >>>>>>>>>>>> +
> >>>>>>>>>>>> + buf->data[0] = buf->base[0];
> >>>>>>>>>>>> + buf->channels = s->channels;
> >>>>>>>>>>>> + buf->nb_samples = pic->nb_samples;
> >>>>>>>>>>>> + buf->sample_fmt = s->sample_fmt;
> >>>>>>>>>>>> + }
> >>>>>>>>>>>> }
> >>>>>>>>>>>> pic->type= FF_BUFFER_TYPE_INTERNAL;
> >>>>>>>>>>>>
> >>>>>>>>>>>> @@ -360,9 +419,15 @@ void avcodec_default_release_buffer(AVCodecContext *s, AVFrame *pic){
> >>>>>>>>>>>> }
> >>>>>>>>>>>> assert(i < s->internal_buffer_count);
> >>>>>>>>>>>> s->internal_buffer_count--;
> >>>>>>>>>>>> + if (buf->base[0] == s->user_buffer) {
> >>>>>>>>>>>> + assert(s->user_buffer_in_use);
> >>>>>>>>>>>> + s->user_buffer_in_use = 0;
> >>>>>>>>>>>> + buf->base[0] = NULL;
> >>>>>>>>>>>> + } else {
> >>>>>>>>>>>> last = &((InternalBuffer*)s->internal_buffer)[s->internal_buffer_count];
> >>>>>>>>>>>>
> >>>>>>>>>>>> FFSWAP(InternalBuffer, *buf, *last);
> >>>>>>>>>>>> + }
> >>>>>>>>>>>>
> >>>>>>>>>>>> for(i=0; i<4; i++){
> >>>>>>>>>>>> pic->data[i]=NULL;
> >>>>>>>>>>> i dont see how this could work
> >>>>>>>>>>> the buffer used and returned by the previous decode() is put in a que by the
> >>>>>>>>>>> user app and user_buffer is set to a new buffer.
> >>>>>>>>>>> also you appear to end up calling av_free()
> >>>>>>>>>>> on user supplied buffers
> >>>>>>>>>> Well, I meant to disallow that, but the documentation I put just says
> >>>>>>>>>> the user cannot free or change the data while user_buffer_in_use is set.
> >>>>>>>>>> I didn't consider the user replacing it with a new buffer. But at any
> >>>>>>>>>> rate, if that should be allowed, things get more complicated. I'll need
> >>>>>>>>>> to add a flag or something to indicate each user-supplied buffer. I'll
> >>>>>>>>>> work on it.
> >>>>>>>>> i think the API is too complex already and i dont see why it is so
> >>>>>>>>> if user buffer is set get_buffer() should return it or fail, if its returned
> >>>>>>>>> it should set user_buffer to NULL
> >>>>>>>>> calling get_buffer() a second time if user_buffer was set should be disallowed
> >>>>>>>>> release_buffer should do nothing
> >>>>>>>>>
> >>>>>>>>> if we ever have decoders that dont work with this then we need a AVCodec flag
> >>>>>>>>> that indicates them. For this case get_buffer() would then ignore user_buffer
> >>>>>>>>> and avcodec_decode() would copy to the provided user_buffer if any.
> >>>>>>>>> (we do not need this currently though because we do not have such a decoder)
> >>>>>>>>>
> >>>>>>>>> maybe iam missing something but this seems simpler
> >>>>>>>> Ok I did it a different way. New patch attached. I'm not 100% sure
> >>>>>>>> about the way reget_buffer() is handled but it works. AVFrame.type
> >>>>>>>> seems to be a mask, but I don't know if it was intended to be used that way.
> >>>>>>>>
> >>>>>>>> Cheers,
> >>>>>>>> Justin
> >>>>>>>>
> >>>>>>>>
> >>>>>>>> doc/APIchanges | 9 ++
> >>>>>>>> libavcodec/avcodec.h | 100 ++++++++++++++++++++++++++++++--
> >>>>>>>> libavcodec/pcm.c | 41 +++++++++++--
> >>>>>>>> libavcodec/utils.c | 157 ++++++++++++++++++++++++++++++++++++++++++++-------
> >>>>>>>> 4 files changed, 275 insertions(+), 32 deletions(-)
> >>>>>>>> 39eb7fb791089c0822e3f87d3226b49131563a72 avcodec_decode_audio4.patch
> >>>>>>>> diff --git a/doc/APIchanges b/doc/APIchanges
> >>>>>>>> index 4155d32..a39d9fd 100644
> >>>>>>>> --- a/doc/APIchanges
> >>>>>>>> +++ b/doc/APIchanges
> >>>>>>>> @@ -13,6 +13,15 @@ libavutil: 2009-03-08
> >>>>>>>>
> >>>>>>>> API changes, most recent first:
> >>>>>>>>
> >>>>>>>> +2010-XX-XX - rXXXXX - lavc 52.92.0 - AVFrame and avcodec_decode_audio
> >>>>>>>> + Add nb_samples field to AVFrame.
> >>>>>>>> + Add user_buffer, user_buffer_size, and user_buffer_in_use fields to AVCodecContext.
> >>>>>>>> + Deprecate AVCODEC_MAX_AUDIO_FRAME_SIZE.
> >>>>>>>> + Deprecate avcodec_decode_audio3() in favor of avcodec_decode_audio4().
> >>>>>>>> + avcodec_decode_audio4() writes output samples to an AVFrame, which gives the
> >>>>>>>> + audio decoders the ability to use get/release/reget_buffer() to get an
> >>>>>>>> + output buffer.
> >>>>>>>> +
> >>>>>>>> 2010-10-10 - r25441 - lavfi 1.49.0 - AVFilterLink.time_base
> >>>>>>>> Add time_base field to AVFilterLink.
> >>>>>>>>
> >>>>>>>> diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
> >>>>>>>> index 4bddbaa..1aa1c8c 100644
> >>>>>>>> --- a/libavcodec/avcodec.h
> >>>>>>>> +++ b/libavcodec/avcodec.h
> >>>>>>>> @@ -31,7 +31,7 @@
> >>>>>>>> #include "libavutil/cpu.h"
> >>>>>>>>
> >>>>>>>> #define LIBAVCODEC_VERSION_MAJOR 52
> >>>>>>>> -#define LIBAVCODEC_VERSION_MINOR 92
> >>>>>>>> +#define LIBAVCODEC_VERSION_MINOR 93
> >>>>>>>> #define LIBAVCODEC_VERSION_MICRO 0
> >>>>>>>>
> >>>>>>>> #define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
> >>>>>>>> @@ -467,8 +467,10 @@ enum SampleFormat {
> >>>>>>>> CH_FRONT_LEFT_OF_CENTER|CH_FRONT_RIGHT_OF_CENTER)
> >>>>>>>> #define CH_LAYOUT_STEREO_DOWNMIX (CH_STEREO_LEFT|CH_STEREO_RIGHT)
> >>>>>>>>
> >>>>>>>> +#if FF_API_AUDIO_OLD
> >>>>>>>> /* in bytes */
> >>>>>>>> #define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000 // 1 second of 48khz 32bit audio
> >>>>>>>> +#endif
> >>>>>>>>
> >>>>>>>> /**
> >>>>>>>> * Required number of additionally allocated bytes at the end of the input bitstream for decoding.
> >>>>>>>> @@ -988,7 +990,13 @@ typedef struct AVPanScan{
> >>>>>>>> * - decoding: Set by libavcodec\
> >>>>>>>> */\
> >>>>>>>> void *hwaccel_picture_private;\
> >>>>>>>> -
> >>>>>>>> +\
> >>>>>>>> + /**\
> >>>>>>>> + * number of audio samples (per channel) described by this frame\
> >>>>>>>> + * - encoding: Set by user.\
> >>>>>>>> + * - decoding: Set by libavcodec.\
> >>>>>>>> + */\
> >>>>>>>> + int nb_samples;\
> >>>>>>>>
> >>>>>>>> #define FF_QSCALE_TYPE_MPEG1 0
> >>>>>>>> #define FF_QSCALE_TYPE_MPEG2 1
> >>>>>>>> @@ -2744,6 +2752,33 @@ typedef struct AVCodecContext {
> >>>>>>>> * - decoding: unused
> >>>>>>>> */
> >>>>>>>> int lpc_passes;
> >>>>>>>> +
> >>>>>>>> + /**
> >>>>>>>> + * User-allocated audio decoder output buffer & buffer size
> >>>>>>>> + * If user_buffer is non-NULL and is large enough,
> >>>>>>>> + * avcodec_default_get_buffer() may user it as an internal buffer instead
> >>>>>>>> + * of allocating its own. This only works with decoders that support
> >>>>>>>> + * CODEC_CAP_DR1. If the decoder uses this buffer, it will set the value
> >>>>>>>> + * to NULL.
> >>>>>>>> + *
> >>>>>>>> + * @note The user may not use this field when using avcodec_decode_audio3()
> >>>>>>>> + * or avcodec_decode_audio2().
> >>>>>>> I dont see why we need such special casing. *samples could be passed in the new
> >>>>>>> api like it is in the old
> >>>>>> The old API already has the samples buffer passed directly. Why should
> >>>>>> the old API be changed to accept user_buffer as an alternative to
> >>>>>> *samples? Do they have to match? If not, which takes priority? This
> >>>>>> would require added documentation to an old API. That seems more
> >>>>>> complexity than necessary when the ability to supply a direct buffer is
> >>>>>> already there.
> >>>>> you misunderstand
> >>>>> why is the new api having it passed over AVCodecContext and the old over an
> >>>>> function argument (in the sense why dont you change the new api to also take
> >>>>> a argument for that?) maybe i miss something but this seems to be simpler
> >>>> Ah, thanks for clarifying.
> >>>>
> >>>> The new API doesn't need it. The old API needs it in order to avoid
> >>>> memcpy. In the new API, if the user wants a direct buffer, she can
> >>>> override get/release_buffer().
> >>> she can but that is more code and work
> >> Ok. I'm somewhat against it, but I'm willing to implement it.
> >
> > can you elaborate on why you are against it?
> > to me this looks nicer and more logic from the outsides users point of view
>
> I would feel better about it if we have something in the documentation
> saying that not passing a direct buffer is the preferred usage since it
> will ensure that a large enough buffer will be allocated as long as
> enough memory is available and the audio parameters are within a sane range.
The documentation can be changed ...
[...]
--
Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
I know you won't believe me, but the highest form of Human Excellence is
to question oneself and others. -- Socrates
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