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ACM: Corrected temporary buffer size

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This CL corrects the temporary buffers size in the
pre-processing of the capture audio before encoding.

As part of this it removes the ACM-specific hardcoding
of the size and instead ensures that the size of the
temporary buffer matches that of the AudioFrame.

Bug: webrtc:11242
Change-Id: I56dd6cadfd4e140e8e159966c33d1027383ea9fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170340
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30775}
This commit is contained in:
Per Åhgren
2020-03-12 11:53:30 +01:00
committed by Commit Bot
parent c71be24c82
commit d82a02c837
3 changed files with 20 additions and 9 deletions
Download Patch File Download Diff File Expand all files Collapse all files

10
modules/audio_coding/test/EncodeDecodeTest.cc
View File

@ -24,6 +24,12 @@
namespace webrtc {
namespace {
// Buffer size for stereo 48 kHz audio.
constexpr size_t kWebRtc10MsPcmAudio = 960;
} // namespace
TestPacketization::TestPacketization(RTPStream* rtpStream, uint16_t frequency)
: _rtpStream(rtpStream), _frequency(frequency), _seqNo(0) {}
@ -92,7 +98,7 @@ void Sender::Run() {
}
Receiver::Receiver()
: _playoutLengthSmpls(WEBRTC_10MS_PCM_AUDIO),
: _playoutLengthSmpls(kWebRtc10MsPcmAudio),
_payloadSizeBytes(MAX_INCOMING_PAYLOAD) {}
void Receiver::Setup(AudioCodingModule* acm,
@ -139,7 +145,7 @@ void Receiver::Setup(AudioCodingModule* acm,
_pcmFile.Open(file_name, 32000, "wb+");
_realPayloadSizeBytes = 0;
_playoutBuffer = new int16_t[WEBRTC_10MS_PCM_AUDIO];
_playoutBuffer = new int16_t[kWebRtc10MsPcmAudio];
_frequency = playSampFreq;
_acm = acm;
_firstTime = true;