Hello,
I am trying to capture the "default" audio output using alsa API and use
ffmpeg to compress the raw stream. I set the captured format to be
SND_PCM_FORMAT_S16_LE and use CODEC_ID_MP2 from ffmpeg to compress the
captured stream. For some reasons, the output stream is corrupted and I can
hear periodic bursting noise in the resulted audio. When I tell ffmpeg to
use CODEC_ID_PCM_S16LE to compress the captured stream, the output stream is
good. Is there any kind of post processing I need to do to the captured
stream before I can compress? Here is roughly what my code looks like:
ret = snd_pcm_open(&handle, "default",
SND_PCM_STREAM_CAPTURE, 0);
snd_pcm_hw_params_alloca(¶ms);
snd_pcm_hw_params_any(handle, params);
snd_pcm_hw_params_set_access(handle, params,
SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_format(handle, params,
SND_PCM_FORMAT_S16_LE);
snd_pcm_hw_params_set_channels(handle, params, 2);
snd_pcm_hw_params_set_rate_near(handle, params,
&val, &dir);
frames = 32;
snd_pcm_hw_params_set_period_size_near(handle,
params, &frames, &dir);
audio_buffer_size = frames * 4; /* 2 bytes/sample, 2 channels */
audio_buffer = av_malloc(audio_buffer_size);
ret = snd_pcm_hw_params(handle, params);
snd_pcm_hw_params_get_period_size(params,
&frames, &dir);
/* .................... */
/* ffmpeg */
audio_st = av_new_stream(formatContext, 1);
if (!audio_st) {
fprintf(stderr, "Could not alloc audio stream\n");
exit(1);
}
pAudioCxt = audio_st->codec;
pAudioCxt->codec_id = pOutputFmt->audio_codec;
pAudioCxt->codec_type = CODEC_TYPE_AUDIO;
/* sample parameters */
pAudioCxt->bit_rate = 64000;
pAudioCxt->sample_rate = 44100;
pAudioCxt->channels = 2;
/* find audio codec */
/*pAudioCodec = avcodec_find_encoder(pAudioCxt->codec_id);*/
/* capture */
ret = snd_pcm_readi(handle, audio_buffer, frames);
/* encode */
out_size = avcodec_encode_audio(pAudioContext, enc_aud_buff,
audio_buffer_size, (short *)audio_buffer);
Thanks,
Mik
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