On Mon, 2016-01-18 at 13:06 +0530, arun at accosted.net wrote:
> This is required to have unequal channel counts on capture in and out
> streams, which is needed for beamforming to work.
The commit message should mention why the sample format was changed to
float.
> Â void pa_webrtc_ec_play(pa_echo_canceller *ec, const uint8_t *play) {
> Â Â Â Â Â webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.webrtc.apm;
> -Â Â Â Â webrtc::AudioFrame play_frame;
> Â Â Â Â Â const pa_sample_spec *ss = &ec->params.webrtc.play_ss;
> +Â Â Â Â int n = ec->params.webrtc.blocksize;
> +Â Â Â Â float **buf = ec->params.webrtc.play_buffer;
> +Â Â Â Â webrtc::StreamConfig config(ss->rate, ss->channels, false);
> Â
> -Â Â Â Â play_frame.num_channels_ = ss->channels;
> -Â Â Â Â play_frame.sample_rate_hz_ = ss->rate;
> -Â Â Â Â play_frame.interleaved_ = true;
> -Â Â Â Â play_frame.samples_per_channel_ = ec->params.webrtc.blocksize;
> -
> -Â Â Â Â pa_assert(play_frame.samples_per_channel_ <= webrtc::AudioFrame::kMaxDataSizeSamples);
> -Â Â Â Â memcpy(play_frame.data_, play, ec->params.webrtc.blocksize * pa_frame_size(ss));
> +Â Â Â Â pa_deinterleave(play, (void **) buf, ss->channels, pa_sample_size(ss), n);
> Â
> -Â Â Â Â if (apm->ProcessReverseStream(&play_frame) != webrtc::AudioProcessing::kNoError)
> +Â Â Â Â if (apm->ProcessReverseStream(buf, config, config, buf) != webrtc::AudioProcessing::kNoError)
> Â Â Â Â Â Â Â Â Â pa_log("Failed to process playback stream");
> +
> +Â Â Â Â /* FIXME: we need to be able to modify playback samples */
This comment should say also why we need to be able to modify them (and
I also don't understand what's preventing us from doing so now).
> @@ -573,4 +575,9 @@ void pa_webrtc_ec_done(pa_echo_canceller *ec) {
> Â Â Â Â Â Â Â Â Â delete (webrtc::AudioProcessing*)ec->params.webrtc.apm;
> Â Â Â Â Â Â Â Â Â ec->params.webrtc.apm = NULL;
> Â Â Â Â Â }
> +
> +Â Â Â Â for (i = 0; i < ec->params.webrtc.rec_ss.channels; i++)
> +Â Â Â Â Â Â Â Â pa_xfree(ec->params.webrtc.rec_buffer[i]);
> +Â Â Â Â for (i = 0; i < ec->params.webrtc.play_ss.channels; i++)
> +Â Â Â Â Â Â Â Â pa_xfree(ec->params.webrtc.play_buffer[i]);
I think it's not guaranteed that rec_ss and play_ss are initialized at
this point. You should check that rec_buffer and play_buffer are non-
NULL before accessing them.
--
Tanu
