Remove the ACMResampler class, use PushResampler directly instead.
Bug: chromium:335805780
Change-Id: Ieaf127046fe79a53c2bbde71d6e08f3ce07cbf09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391280
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44698}
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index d279433..dd4500f 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -760,6 +760,7 @@
"../../api/neteq:default_neteq_factory",
"../../api/neteq:neteq_api",
"../../api/units:timestamp",
+ "../../common_audio",
"../../rtc_base:checks",
"../../rtc_base:digest",
"../../rtc_base:stringutils",
diff --git a/modules/audio_coding/acm2/acm_resampler.cc b/modules/audio_coding/acm2/acm_resampler.cc
index 1587ecc..9daa04d 100644
--- a/modules/audio_coding/acm2/acm_resampler.cc
+++ b/modules/audio_coding/acm2/acm_resampler.cc
@@ -10,51 +10,16 @@
#include "modules/audio_coding/acm2/acm_resampler.h"
-#include <string.h>
-
+#include <array>
#include <cstdint>
#include "api/audio/audio_frame.h"
#include "api/audio/audio_view.h"
#include "rtc_base/checks.h"
-#include "rtc_base/logging.h"
-#include "rtc_base/numerics/safe_conversions.h"
namespace webrtc {
namespace acm2 {
-ACMResampler::ACMResampler() {}
-
-ACMResampler::~ACMResampler() {}
-
-int ACMResampler::Resample10Msec(const int16_t* in_audio,
- int in_freq_hz,
- int out_freq_hz,
- size_t num_audio_channels,
- size_t out_capacity_samples,
- int16_t* out_audio) {
- InterleavedView<const int16_t> src(
- in_audio, SampleRateToDefaultChannelSize(in_freq_hz), num_audio_channels);
- InterleavedView<int16_t> dst(out_audio,
- SampleRateToDefaultChannelSize(out_freq_hz),
- num_audio_channels);
- RTC_DCHECK_GE(out_capacity_samples, dst.size());
- Resample10Msec(src, in_freq_hz, dst, out_freq_hz);
- return dst.samples_per_channel();
-}
-
-void ACMResampler::Resample10Msec(InterleavedView<const int16_t> src,
- int in_freq_hz,
- InterleavedView<int16_t> dst,
- int out_freq_hz) {
- RTC_DCHECK_EQ(src.num_channels(), dst.num_channels());
- if (in_freq_hz == out_freq_hz) {
- CopySamples(dst, src);
- } else {
- resampler_.Resample(src, dst);
- }
-}
-
ResamplerHelper::ResamplerHelper() {
ClearSamples(last_audio_buffer_);
}
@@ -63,44 +28,38 @@
AudioFrame* audio_frame) {
const int current_sample_rate_hz = audio_frame->sample_rate_hz_;
RTC_DCHECK_NE(current_sample_rate_hz, 0);
+ RTC_DCHECK_GT(desired_sample_rate_hz, 0);
// Update if resampling is required.
+ // TODO(tommi): `desired_sample_rate_hz` should never be -1.
+ // Remove the first check.
const bool need_resampling =
(desired_sample_rate_hz != -1) &&
(current_sample_rate_hz != desired_sample_rate_hz);
if (need_resampling && !resampled_last_output_frame_) {
// Prime the resampler with the last frame.
- int16_t temp_output[AudioFrame::kMaxDataSizeSamples];
- int samples_per_channel_int = resampler_.Resample10Msec(
- last_audio_buffer_.data(), current_sample_rate_hz,
- desired_sample_rate_hz, audio_frame->num_channels_,
- AudioFrame::kMaxDataSizeSamples, temp_output);
- if (samples_per_channel_int < 0) {
- RTC_LOG(LS_ERROR) << "AcmReceiver::GetAudio - "
- "Resampling last_audio_buffer_ failed.";
- return false;
- }
+ InterleavedView<const int16_t> src(last_audio_buffer_.data(),
+ audio_frame->samples_per_channel(),
+ audio_frame->num_channels());
+ std::array<int16_t, AudioFrame::kMaxDataSizeSamples> temp_output;
+ InterleavedView<int16_t> dst(
+ temp_output.data(),
+ SampleRateToDefaultChannelSize(desired_sample_rate_hz),
+ audio_frame->num_channels_);
+ resampler_.Resample(src, dst);
}
// TODO(bugs.webrtc.org/3923) Glitches in the output may appear if the output
// rate from NetEq changes.
if (need_resampling) {
- // TODO(yujo): handle this more efficiently for muted frames.
- int samples_per_channel_int = resampler_.Resample10Msec(
- audio_frame->data(), current_sample_rate_hz, desired_sample_rate_hz,
- audio_frame->num_channels_, AudioFrame::kMaxDataSizeSamples,
- audio_frame->mutable_data());
- if (samples_per_channel_int < 0) {
- RTC_LOG(LS_ERROR)
- << "AcmReceiver::GetAudio - Resampling audio_buffer_ failed.";
- return false;
- }
- audio_frame->samples_per_channel_ =
- static_cast<size_t>(samples_per_channel_int);
- audio_frame->sample_rate_hz_ = desired_sample_rate_hz;
- RTC_DCHECK_EQ(audio_frame->sample_rate_hz_,
- dchecked_cast<int>(audio_frame->samples_per_channel_ * 100));
+ // Grab the source view of the current layout before changing properties.
+ InterleavedView<const int16_t> src = audio_frame->data_view();
+ audio_frame->SetSampleRateAndChannelSize(desired_sample_rate_hz);
+ InterleavedView<int16_t> dst = audio_frame->mutable_data(
+ audio_frame->samples_per_channel(), audio_frame->num_channels());
+ // TODO(tommi): Don't resample muted audio frames.
+ resampler_.Resample(src, dst);
resampled_last_output_frame_ = true;
} else {
resampled_last_output_frame_ = false;
@@ -108,10 +67,10 @@
}
// Store current audio in `last_audio_buffer_` for next time.
- // TODO: b/335805780 - Use CopySamples().
- memcpy(last_audio_buffer_.data(), audio_frame->data(),
- sizeof(int16_t) * audio_frame->samples_per_channel_ *
- audio_frame->num_channels_);
+ InterleavedView<int16_t> dst(last_audio_buffer_.data(),
+ audio_frame->samples_per_channel(),
+ audio_frame->num_channels());
+ CopySamples(dst, audio_frame->data_view());
return true;
}
diff --git a/modules/audio_coding/acm2/acm_resampler.h b/modules/audio_coding/acm2/acm_resampler.h
index 256884b..8b6cee0 100644
--- a/modules/audio_coding/acm2/acm_resampler.h
+++ b/modules/audio_coding/acm2/acm_resampler.h
@@ -17,34 +17,11 @@
#include <array>
#include "api/audio/audio_frame.h"
-#include "api/audio/audio_view.h"
#include "common_audio/resampler/include/push_resampler.h"
namespace webrtc {
namespace acm2 {
-class ACMResampler {
- public:
- ACMResampler();
- ~ACMResampler();
-
- // TODO: b/335805780 - Remove and use only the InterleavedView<> version
- int Resample10Msec(const int16_t* in_audio,
- int in_freq_hz,
- int out_freq_hz,
- size_t num_audio_channels,
- size_t out_capacity_samples,
- int16_t* out_audio);
-
- void Resample10Msec(InterleavedView<const int16_t> src,
- int src_freq_hz,
- InterleavedView<int16_t> dst,
- int dst_freq_hz);
-
- private:
- PushResampler<int16_t> resampler_;
-};
-
// Helper class to perform resampling if needed, meant to be used after
// receiving the audio_frame from NetEq. Provides reasonably glitch free
// transitions between different output sample rates from NetEq.
@@ -56,7 +33,7 @@
bool MaybeResample(int desired_sample_rate_hz, AudioFrame* audio_frame);
private:
- ACMResampler resampler_;
+ PushResampler<int16_t> resampler_;
bool resampled_last_output_frame_ = true;
std::array<int16_t, AudioFrame::kMaxDataSizeSamples> last_audio_buffer_;
};
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index 32a36a7..41d022e 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -23,8 +23,8 @@
#include "api/audio/audio_view.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/function_view.h"
+#include "common_audio/resampler/include/push_resampler.h"
#include "modules/audio_coding/acm2/acm_remixing.h"
-#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "modules/include/module_common_types_public.h"
#include "rtc_base/buffer.h"
@@ -161,7 +161,7 @@
Buffer encode_buffer_ RTC_GUARDED_BY(acm_mutex_);
uint32_t expected_codec_ts_ RTC_GUARDED_BY(acm_mutex_);
uint32_t expected_in_ts_ RTC_GUARDED_BY(acm_mutex_);
- acm2::ACMResampler resampler_ RTC_GUARDED_BY(acm_mutex_);
+ PushResampler<int16_t> resampler_ RTC_GUARDED_BY(acm_mutex_);
ChangeLogger bitrate_logger_ RTC_GUARDED_BY(acm_mutex_);
// Current encoder stack, provided by a call to RegisterEncoder.
@@ -506,11 +506,10 @@
preprocess_frame_.SetSampleRateAndChannelSize(encoder_stack_->SampleRateHz());
if (resample) {
- resampler_.Resample10Msec(
- resample_src_audio, in_frame.sample_rate_hz(),
+ resampler_.Resample(
+ resample_src_audio,
preprocess_frame_.mutable_data(preprocess_frame_.samples_per_channel(),
- preprocess_frame_.num_channels()),
- preprocess_frame_.sample_rate_hz());
+ preprocess_frame_.num_channels()));
}
expected_codec_ts_ +=
diff --git a/modules/audio_coding/test/opus_test.cc b/modules/audio_coding/test/opus_test.cc
index da44f26..6b511c2 100644
--- a/modules/audio_coding/test/opus_test.cc
+++ b/modules/audio_coding/test/opus_test.cc
@@ -267,9 +267,8 @@
// If input audio is sampled at 32 kHz, resampling to 48 kHz is required.
InterleavedView<int16_t> dst(&audio[written_samples], 480, channels);
- resampler_.Resample10Msec(audio_frame.data_view(),
- audio_frame.sample_rate_hz_, dst, 48000);
- written_samples += 480 * channels;
+ resampler_.Resample(audio_frame.data_view(), dst);
+ written_samples += dst.size();
// Sometimes we need to loop over the audio vector to produce the right
// number of packets.
diff --git a/modules/audio_coding/test/opus_test.h b/modules/audio_coding/test/opus_test.h
index d6f0e7c..4d2586e 100644
--- a/modules/audio_coding/test/opus_test.h
+++ b/modules/audio_coding/test/opus_test.h
@@ -18,6 +18,7 @@
#include <memory>
#include "api/neteq/neteq.h"
+#include "common_audio/resampler/include/push_resampler.h"
#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/codecs/opus/opus_inst.h"
#include "modules/audio_coding/test/PCMFile.h"
@@ -51,7 +52,7 @@
int counter_;
uint8_t payload_type_;
uint32_t rtp_timestamp_;
- acm2::ACMResampler resampler_;
+ PushResampler<int16_t> resampler_;
WebRtcOpusEncInst* opus_mono_encoder_;
WebRtcOpusEncInst* opus_stereo_encoder_;
WebRtcOpusDecInst* opus_mono_decoder_;
