Hi I finally decided to post here to see if anybody can help me. I have written an audio decoder application that streams audio packets from a file (mostly mp3s), decodes and renders via ALSA. I setup the alsa hw in interleaved mode, 2 channel (mp3 has two channels of audio), submit decoded frames while keeping the buffer optimal in a play thread. But what I hear is destorted audio, like the sound coming from an old cassette player when its playing and you hit the fast forward button. However when I convert the mp3 into one channel with ffmpeg, everything plays fine, sound comes out from two speakers. I am not sure if I understand what's going on. Since I set the device in interleaved mode and I am providing the channel data as they are embedded inside the file, do I need to do anything additional? I am using ffmpeg streming APIs, I dont know if a particular channels data has to be sent to a particular channel, besides, how would I do that? I am using snd_pcm_writei, and there is no channel destination in the API. BTW the data is packed in 16 bits boundary. if anybody have any clue, please let me know. Thanks Ratin ------------------------------------------------------------------------------ Site24x7 APM Insight: Get Deep Visibility into Application Performance APM + Mobile APM + RUM: Monitor 3 App instances at just $35/Month Monitor end-to-end web transactions and take corrective actions now Troubleshoot faster and improve end-user experience. Signup Now! http://pubads.g.doubleclick.net/gampad/clk?id=267308311&iu=/4140 _______________________________________________ Alsa-user mailing list Alsa-user@xxxxxxxxxxxxxxxxxxxxx https://lists.sourceforge.net/lists/listinfo/alsa-user
