[Libav-user] AudioToolbox AudioBufferList to/from AVFilter

Paul B Mahol onemda at gmail.com
Tue Nov 28 22:28:08 EET 2017


On 11/28/17, Ronak <ronak2121-at-yahoo.com at ffmpeg.org> wrote:
> Hey,
>
> Yes I have been going through the examples, and I am getting an EAGAIN
> error. I'd like to find out why and what I have to do.
>
> The samples are not really clear about what I have to do.

EAGAIN errors are OK, that just means you need to provode more data
before receiving any.

>
> Ronak
>
>> On Nov 28, 2017, at 3:21 PM, Paul B Mahol <onemda at gmail.com> wrote:
>>
>> On 11/28/17, Ronak <ronak2121-at-yahoo.com at ffmpeg.org> wrote:
>>> Hi Paul,
>>>
>>> Thanks for that, that fixed the problem. Now, I'm getting -35 errors when
>>> I
>>> try to read the audio out of the buffer sink:
>>>
>>> Why would that happen?
>>>
>>> - (void)allocateRenderResourcesAudioFormat:(AVAudioFormat *
>>> _Nonnull)format
>>> capacity:(AVAudioFrameCount __unused)frameCapacity {
>>>
>>>  NSString *bufferArgs = [[NSString alloc]
>>> initWithFormat:@"sample_rate=%f:channels=%d:sample_fmt=%s:channel_layout=%d",
>>> format.sampleRate, format.channelCount,
>>> av_get_sample_fmt_name(AV_SAMPLE_FMT_FLTP), AV_CH_LAYOUT_STEREO];
>>>
>>>  avfilter_graph_create_filter(&_bufferContext,
>>> avfilter_get_by_name("abuffer"), "buffer_context", bufferArgs.UTF8String,
>>> NULL, self.filterGraph);
>>>  avfilter_graph_create_filter(&_bufferSinkContext,
>>> avfilter_get_by_name("abuffersink"), "buffer_sink", NULL, NULL,
>>> self.filterGraph);
>>>
>>>  avfilter_graph_create_filter(&_bassFilterContext,
>>> avfilter_get_by_name("bass"), "bass",
>>> "gain=0:frequency=100:width_type=o:width=1", NULL, self.filterGraph);
>>>  avfilter_graph_create_filter(&_trebleFilterContext,
>>> avfilter_get_by_name("treble"), "treble",
>>> "gain=0:frequency=10000:width_type=o:width=1", NULL, self.filterGraph);
>>>  avfilter_graph_create_filter(&_equalizerFilterContext,
>>> avfilter_get_by_name("equalizer"), "equalizer",
>>> "gain=0:frequency=250:width_type=o:width=1", NULL, self.filterGraph);
>>>
>>>  avfilter_link(_bufferContext, 0, _bassFilterContext, 0);
>>>  avfilter_link(_bassFilterContext, 0, _trebleFilterContext, 0);
>>>  avfilter_link(_trebleFilterContext, 0, _equalizerFilterContext, 0);
>>>  avfilter_link(_equalizerFilterContext, 0, _bufferSinkContext, 0);
>>>
>>>  avfilter_graph_config(self.filterGraph, NULL);
>>> }
>>>
>>> - (void)processBuffer:(AudioBufferList * _Nonnull)buffer
>>> outputBuffer:(AudioBufferList * _Nonnull)outputBuffer {
>>>
>>>  AVFrame *audioFrame = av_frame_alloc();
>>>  audioFrame->channels = 2;
>>>  audioFrame->channel_layout = AV_CH_LAYOUT_STEREO;
>>>  audioFrame->sample_rate = 44100.000000;
>>>  audioFrame->format = AV_SAMPLE_FMT_FLTP;
>>>  audioFrame->nb_samples = buffer->mBuffers[0].mDataByteSize/
>>> sizeof(Float32) * 44100;
>>>  audioFrame->pts = audioFrame->nb_samples;
>>>
>>>  audioFrame->extended_data[0] = buffer->mBuffers[0].mData;
>>>  audioFrame->extended_data[1] = buffer->mBuffers[1].mData;
>>>  audioFrame->linesize[0] = buffer->mBuffers[0].mDataByteSize;
>>>
>>>  int result = av_buffersrc_write_frame(self.bufferContext, audioFrame);
>>>  if (result > 0) {
>>>    AVFrame *returnedFrame = av_frame_alloc();
>>>    int result3 = av_buffersink_get_frame(self.bufferSinkContext,
>>> returnedFrame);
>>>
>>>    NSString *string = [[NSString alloc]
>>> initWithCString:av_err2str(result3)
>>> encoding:NSUTF8StringEncoding]; <---- This shows a -35 error code
>>>    NSLog(@"The string is %@", string);
>>>
>>>    outputBuffer->mBuffers[0].mData = returnedFrame->extended_data[0];
>>>    outputBuffer->mBuffers[1].mData = returnedFrame->extended_data[1];
>>>  } else {
>>>    NSString *string = [[NSString alloc]
>>> initWithCString:av_err2str(result)
>>> encoding:NSUTF8StringEncoding];
>>>    NSLog(@"The string is %@", string);
>>>
>>>    outputBuffer->mBuffers[0].mData = buffer->mBuffers[0].mData;
>>>    outputBuffer->mBuffers[1].mData = buffer->mBuffers[1].mData;
>>>  }
>>> }
>>>
>>> Is there something wrong with the frame I'm passing into the call to
>>> av_buffersink_get_frame?
>>
>> Check that return value is not EOF or EAGAIN, there are simple
>> examples in ffmpeg source tree.
>>
>>>
>>>
>>> This is just something simple I'm trying to get up and running, before I
>>> write production level code.
>>>
>>> Thanks for the help!
>>>
>>> Ronak
>>>
>>>> On Nov 28, 2017, at 11:58 AM, Paul B Mahol <onemda at gmail.com> wrote:
>>>>
>>>> On 11/28/17, Ronak <ronak2121-at-yahoo.com at ffmpeg.org> wrote:
>>>>> I managed to trace this down to av_frame_get_buffer returning -22.
>>>>>
>>>>> Here's the code that I tried:
>>>>>
>>>>> AVFrame *audioFrame = av_frame_alloc();
>>>>> audioFrame->channels = 2;
>>>>> audioFrame->channel_layout = av_get_default_channel_layout(2);
>>>>> audioFrame->sample_rate = 44100;
>>>>> audioFrame->nb_samples = buffer->mBuffers[0].mDataByteSize/
>>>>> sizeof(Float32) * 44100;
>>>>> audioFrame->pts = audioFrame->nb_samples;
>>>>> av_frame_get_buffer(audioFrame, 0); <--- returns -22
>>>>
>>>> You nowhere set sample format.
>>>>
>>>>>
>>>>> audioFrame->extended_data[0] = buffer->mBuffers[0].mData;
>>>>> audioFrame->extended_data[1] = buffer->mBuffers[1].mData;
>>>>> audioFrame->linesize[0] = buffer->mBuffers[0].mDataByteSize;
>>>>>
>>>>> AVFrame *otherFrame = av_frame_alloc();
>>>>> int result2 = av_frame_ref(otherFrame, audioFrame); <--- returns -22
>>>>>
>>>>> int result = av_buffersrc_write_frame(self.bufferContext, audioFrame);
>>>>>
>>>>> Why would av_frame_get_buffer return -22? Am I not supposed to call it?
>>>>> What
>>>>> about write frame?
>>>>>
>>>>>
>>>>>> On Nov 27, 2017, at 7:19 PM, Ronak Patel
>>>>>> <ronak2121-at-yahoo.com at ffmpeg.org> wrote:
>>>>>>
>>>>>> Hi Paul,
>>>>>>
>>>>>> Do you mind pointing me to the relevant documentation?
>>>>>>
>>>>>> I tried setting up an AVFrame instance with the sample rate, channel
>>>>>> layout and data but the calls to av_frame_ref are failing with -22
>>>>>> errors.
>>>>>> I'm looking for any sample code that shows how to properly initialize
>>>>>> an
>>>>>> AVFrame from an AudioBufferList.
>>>>>>
>>>>>> Thanks
>>>>>>
>>>>>> Ronak
>>>>>>
>>>>>> Sent from my iPhone
>>>>>>
>>>>>>> On Nov 26, 2017, at 2:17 PM, Paul B Mahol <onemda at gmail.com> wrote:
>>>>>>>
>>>>>>>> On 11/26/17, Ronak <ronak2121-at-yahoo.com at ffmpeg.org> wrote:
>>>>>>>> Hi,
>>>>>>>>
>>>>>>>> I'm trying to build a graphic equalizer using the ffmpeg library for
>>>>>>>> iOS,
>>>>>>>> wrapping the AVFilter library in an AUAudioUnit.
>>>>>>>>
>>>>>>>> I'm having trouble figuring out how to convert an AudioBufferList's
>>>>>>>> data
>>>>>>>> to
>>>>>>>> an AVFilter and back. The input buffers are in stereo, so I'm also
>>>>>>>> unsure
>>>>>>>> how to pass in both data arrays.
>>>>>>>>
>>>>>>>> Does anyone know how to do this?
>>>>>>>
>>>>>>> Have you read already available documentation?
>>>>>>>
>>>>>>> AVFrame stores samples for packed format into AVFrame->data[0].
>>>>>>> And planar format into AVFrame->extended_data[ X ], where X is
>>>>>>> channel
>>>>>>> number.
>>>>>>> _______________________________________________
>>>>>>> Libav-user mailing list
>>>>>>> Libav-user at ffmpeg.org
>>>>>>> http://ffmpeg.org/mailman/listinfo/libav-user
>>>>>>
>>>>>> _______________________________________________
>>>>>> Libav-user mailing list
>>>>>> Libav-user at ffmpeg.org
>>>>>> http://ffmpeg.org/mailman/listinfo/libav-user
>>>>>
>>>>>
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>>>
>>>
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