I have 2 contexts defined and I and running the samples thru swresample before
sending to AAC encoder. Bur I must have something define wrong as not
interleaved I get only a tone.
Using the following code the define the RAW, PCM and resample contexts.
int status=0;
// setup RAW codec and context
AVCodec *raw_codec = avcodec_find_encoder(AV_CODEC_ID_PCM_S32LE);
if (!raw_codec)
{
log_error("PCM_S32_LE codec id not found!");
return -1;
}
actx->rawctx = avcodec_alloc_context3(raw_codec);
if (!actx->rawctx)
{
log_error("Could not alloc RAW context");
return -1;
}
actx->rawctx->channels = 2;
actx->rawctx->channel_layout = av_get_default_channel_layout(2);
actx->rawctx->sample_rate = DEFAULT_SPEED;
actx->rawctx->sample_fmt = raw_codec->sample_fmts[0]; //
AV_SAMPLE_FMT_S32
actx->rawctx->bit_rate = 2822400; // or 64000
actx->rawctx->time_base.num = 1;
actx->rawctx->time_base.den = 44100;
actx->rawctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL; // Allow
the use of the experimental AAC encoder.
// setup AAC codec and stream context
AVCodec *aac_codec = avcodec_find_encoder(AV_CODEC_ID_AAC);
if (!aac_codec)
{
log_error("AAC codec id not found!");
return -1;
}
actx->audctx = avcodec_alloc_context3(aac_codec);
log_debug("encode aac contet %p", actx->audctx);
if (!actx->audctx)
{
log_error("Could not alloc an encoding context");
return -1;
}
actx->audctx->channels = 2; //use default channels???
actx->audctx->channel_layout = av_get_default_channel_layout(2); //use
default channels???
actx->audctx->sample_rate = 44100;
actx->audctx->sample_fmt = aac_codec->sample_fmts[0];
actx->audctx->bit_rate = 64000;
actx->audctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL; // Allow
the use of the experimental AAC encoder.
if ((status = avcodec_open2(actx->audctx, aac_codec, NULL) < 0))
{
log_error("Could not open output codec (error '%s')", av_err2str(status));
return -1;
}
// setup resampler context
actx->swrctx = swr_alloc_set_opts(NULL,
av_get_default_channel_layout(actx->audctx->channels), actx->audctx->sample_fmt,
actx->audctx->sample_rate,
av_get_default_channel_layout(actx->rawctx->channels), actx->rawctx->sample_fmt,
actx->rawctx->sample_rate, 0, NULL);
if (!actx->swrctx)
{
log_error("Could not allocate resample context");
return -1;
}
if ((status = swr_init(actx->swrctx)) < 0)
{
log_error("Could not open resample context");
swr_free(&actx->swrctx);
return -1;
}
________________________________
From: Libav-user <[email protected]> on behalf of Paul B Mahol
<[email protected]>
Sent: Thursday, November 25, 2021 2:03 AM
To: This list is about using libavcodec, libavformat, libavutil, libavdevice
and libavfilter. <[email protected]>
Subject: Re: [Libav-user] PCM interleaved audio
On Thu, Nov 25, 2021 at 3:38 AM william keeling
<[email protected]<mailto:[email protected]>> wrote:
Does anyone have an example of encoding PCM interleaved audio stream?
I can encode the stream if I manual un-interleave the PCM samples and put them
in R/L planes. I am using “avcodec_find_encoder(AV_CODEC_ID_PCM_S32LE_PLANAR)”
for my manually un-interleaved stream and
“avcodec_find_encoder(AV_CODEC_ID_PCM_S32LE)” to the native interleaved stream.
I am not sure what else I need to do to define the stream as 2 channel
interleaved stream.
For converting between interleaved vs noninterleaved it can be done manually or
via libraries like libswresample of libavfilter.
Thanks
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