On Mon, 2016-01-18 at 13:06 +0530, a...@accosted.net wrote:
> From: Arun Raghavan <g...@arunraghavan.net>
>
> Needed for upcoming beamforming code.
> ---
> src/modules/echo-cancel/echo-cancel.h | 2 +-
> src/modules/echo-cancel/webrtc.cc | 14 ++++++++------
> 2 files changed, 9 insertions(+), 7 deletions(-)
>
> diff --git a/src/modules/echo-cancel/echo-cancel.h
> b/src/modules/echo-cancel/echo-cancel.h
> index 4693516..613f7e3 100644
> --- a/src/modules/echo-cancel/echo-cancel.h
> +++ b/src/modules/echo-cancel/echo-cancel.h
> @@ -65,7 +65,7 @@ struct pa_echo_canceller_params {
> * to C++ linkage. apm is a pointer to an AudioProcessing object
> */
> void *apm;
> int32_t blocksize; /* in frames */
> - pa_sample_spec rec_ss, play_ss;
> + pa_sample_spec rec_ss, play_ss, out_ss;
> bool agc;
> bool trace;
> bool first;
> diff --git a/src/modules/echo-cancel/webrtc.cc
> b/src/modules/echo-cancel/webrtc.cc
> index 5741f45..5f00286 100644
> --- a/src/modules/echo-cancel/webrtc.cc
> +++ b/src/modules/echo-cancel/webrtc.cc
> @@ -333,6 +333,7 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
> ec->params.webrtc.apm = apm;
> ec->params.webrtc.rec_ss = *rec_ss;
> ec->params.webrtc.play_ss = *play_ss;
> + ec->params.webrtc.out_ss = *out_ss;
> ec->params.webrtc.blocksize =
> (uint64_t) (pa_bytes_per_second(out_ss) / pa_frame_size(out_ss)) *
> BLOCK_SIZE_US / PA_USEC_PER_SEC;
> *nframes = ec->params.webrtc.blocksize;
> @@ -372,17 +373,18 @@ void pa_webrtc_ec_play(pa_echo_canceller *ec, const
> uint8_t *play) {
> void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t
> *out) {
> webrtc::AudioProcessing *apm =
> (webrtc::AudioProcessing*)ec->params.webrtc.apm;
> webrtc::AudioFrame out_frame;
> - const pa_sample_spec *ss = &ec->params.webrtc.rec_ss;
> + const pa_sample_spec *rec_ss = &ec->params.webrtc.rec_ss;
> + const pa_sample_spec *out_ss = &ec->params.webrtc.out_ss;
> pa_cvolume v;
> int old_volume, new_volume;
>
> - out_frame.num_channels_ = ss->channels;
> - out_frame.sample_rate_hz_ = ss->rate;
> + out_frame.num_channels_ = rec_ss->channels;
> + out_frame.sample_rate_hz_ = rec_ss->rate;
> out_frame.interleaved_ = true;
> out_frame.samples_per_channel_ = ec->params.webrtc.blocksize;
>
> pa_assert(out_frame.samples_per_channel_ <=
> webrtc::AudioFrame::kMaxDataSizeSamples);
Not directly related to this patch, but this assertion seems wrong. We
compare kMaxDataSizeSamples to the number of frames, but
kMaxDataSizeSamples is calculated as frames * channels.
Also, kMaxDataSizeSamples assumes that the maximum sample spec is 16-
bit, stereo, 32 kHz, and that the maximum buffer size is 60 ms.
However, we seem to allow using 48 kHz, and I didn't find any
limitations for the channel count. We seem to use 10 ms buffers, so
that gives some headroom. webrtc_ec_fixate_spec() should check all
this, and make sure that our parameters don't exceed
kMaxDataSizeSamples.
The changes in this patch all look good.
--
Tanu
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