[Libav-user] AudioToolbox AudioBufferList to/from AVFilter
Ronak
ronak2121 at yahoo.com
Tue Nov 28 22:25:22 EET 2017
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.
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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