[Merge to M80] - ACM: Corrected temporary buffer size
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.
TBR=kwiberg@webrtc.org
(cherry picked from commit d82a02c837d33cdfd75121e40dcccd32515e42d6)
No-Try: True
Bug: webrtc:11242, chromium:1060647
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-Original-Commit-Position: refs/heads/master@{#30775}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171582
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/branch-heads/3987@{#8}
Cr-Branched-From: 1256d9bcac500d962e884231b0360d8c3eb3ef02-refs/heads/master@{#30022}
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index b68579b..ee31900 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -38,6 +38,8 @@
// 48 kHz data.
constexpr size_t kInitialInputDataBufferSize = 6 * 480;
+constexpr int32_t kMaxInputSampleRateHz = 192000;
+
class AudioCodingModuleImpl final : public AudioCodingModule {
public:
explicit AudioCodingModuleImpl(const AudioCodingModule::Config& config);
@@ -348,7 +350,7 @@
return -1;
}
- if (audio_frame.sample_rate_hz_ > 192000) {
+ if (audio_frame.sample_rate_hz_ > kMaxInputSampleRateHz) {
assert(false);
RTC_LOG(LS_ERROR) << "Cannot Add 10 ms audio, input frequency not valid";
return -1;
@@ -465,20 +467,25 @@
*ptr_out = &preprocess_frame_;
preprocess_frame_.num_channels_ = in_frame.num_channels_;
preprocess_frame_.samples_per_channel_ = in_frame.samples_per_channel_;
- std::array<int16_t, WEBRTC_10MS_PCM_AUDIO> audio;
- const int16_t* src_ptr_audio = in_frame.data();
+ std::array<int16_t, AudioFrame::kMaxDataSizeSamples> audio;
+ const int16_t* src_ptr_audio;
if (down_mix) {
- // If a resampling is required the output of a down-mix is written into a
+ // If a resampling is required, the output of a down-mix is written into a
// local buffer, otherwise, it will be written to the output frame.
int16_t* dest_ptr_audio =
resample ? audio.data() : preprocess_frame_.mutable_data();
+ RTC_DCHECK_GE(audio.size(), preprocess_frame_.samples_per_channel_);
RTC_DCHECK_GE(audio.size(), in_frame.samples_per_channel_);
DownMixFrame(in_frame,
rtc::ArrayView<int16_t>(
dest_ptr_audio, preprocess_frame_.samples_per_channel_));
preprocess_frame_.num_channels_ = 1;
- // Set the input of the resampler is the down-mixed signal.
+
+ // Set the input of the resampler to the down-mixed signal.
src_ptr_audio = audio.data();
+ } else {
+ // Set the input of the resampler to the original data.
+ src_ptr_audio = in_frame.data();
}
preprocess_frame_.timestamp_ = expected_codec_ts_;
diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index d8c9260..4d75172 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -33,8 +33,6 @@
class AudioFrame;
struct RTPHeader;
-#define WEBRTC_10MS_PCM_AUDIO 960 // 16 bits super wideband 48 kHz
-
// Callback class used for sending data ready to be packetized
class AudioPacketizationCallback {
public:
diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc
index 20e415d..7633dc2 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -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) {}
@@ -91,7 +97,7 @@
}
Receiver::Receiver()
- : _playoutLengthSmpls(WEBRTC_10MS_PCM_AUDIO),
+ : _playoutLengthSmpls(kWebRtc10MsPcmAudio),
_payloadSizeBytes(MAX_INCOMING_PAYLOAD) {}
void Receiver::Setup(AudioCodingModule* acm,
@@ -138,7 +144,7 @@
_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;
