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Consolidate audio conversion from Channel and TransmitMixer.

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Replace the two versions with a single DownConvertToCodecFormat. As
mentioned in comments, this could be further consolidated with
RemixAndResample but we should write a full audio converter class in
that case.

Along the way:
- Fix the bug present in Channel::Demultiplex with mono input and a
stereo codec.
- Remove the 32 kHz max from the OnDataAvailable path. This avoids a
48 -> 32 -> 48 conversion when VoE is passed 48 kHz audio; instead we
get a straight pass-through to ACM. The 32 kHz conversion is still
needed in the RecordedDataIsAvailable path until APM natively supports
48 kHz.
- Merge resampler improvements from ACM1 to ACM2. This allows ACM to
handle 44.1 kHz audio passed to VoE and was originally done here:
https://webrtc-codereview.appspot.com/1590004
- Reuse the RemixAndResample unit tests for DownConvertToCodecFormat.
- Remove unused functions from utility.cc.

BUG=3155,3000,b/12867572
TESTED=voe_cmd_test using both the OnDataAvailable and
RecordedDataIsAvailable paths, with a captured audio format of all
combinations of {44.1,48} kHz and {1,2} channels, running through all
codecs, and finally using both ACM1 and ACM2.

R=henrika@webrtc.org, turaj@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11019005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5843 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org
2014-04-03 21:56:01 +00:00
parent cca888a5bf
commit 40ee3d07ed
18 changed files with 1663 additions and 1791 deletions
Download Patch File Download Diff File Expand all files Collapse all files

8
webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
View File

@ -1273,11 +1273,7 @@ int32_t AudioCodingModuleImpl::Add10MsData(
return -1;
}
// Allow for 8, 16, 32 and 48kHz input audio.
if ((audio_frame.sample_rate_hz_ != 8000)
&& (audio_frame.sample_rate_hz_ != 16000)
&& (audio_frame.sample_rate_hz_ != 32000)
&& (audio_frame.sample_rate_hz_ != 48000)) {
if (audio_frame.sample_rate_hz_ > 48000) {
assert(false);
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
"Cannot Add 10 ms audio, input frequency not valid");
@ -1444,7 +1440,7 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
if (preprocess_frame_.samples_per_channel_ < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
"Cannot add 10 ms audio, resmapling failed");
"Cannot add 10 ms audio, resampling failed");
return -1;
}
preprocess_frame_.sample_rate_hz_ = send_codec_inst_.plfreq;