Fix the maximum native sample rate in AudioProcessing
BUG=webrtc:4983
R=andrew@webrtc.org, henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1338833002 .
Cr-Commit-Position: refs/heads/master@{#10037}
diff --git a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
index da9d1fa..82480d2 100644
--- a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
+++ b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
@@ -290,8 +290,9 @@
// We only use the limiter if it supports the output sample rate and
// we're actually mixing multiple streams.
- use_limiter_ = _numMixedParticipants > 1 &&
- _outputFrequency <= kAudioProcMaxNativeSampleRateHz;
+ use_limiter_ =
+ _numMixedParticipants > 1 &&
+ _outputFrequency <= AudioProcessing::kMaxNativeSampleRateHz;
MixFromList(mixedAudio, mixList);
MixAnonomouslyFromList(mixedAudio, additionalFramesList);
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index ff4128b..4ef4e6d 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -147,6 +147,17 @@
int volume_;
};
+const int AudioProcessing::kNativeSampleRatesHz[] = {
+ AudioProcessing::kSampleRate8kHz,
+ AudioProcessing::kSampleRate16kHz,
+ AudioProcessing::kSampleRate32kHz,
+ AudioProcessing::kSampleRate48kHz};
+const size_t AudioProcessing::kNumNativeSampleRates =
+ arraysize(AudioProcessing::kNativeSampleRatesHz);
+const int AudioProcessing::kMaxNativeSampleRateHz = AudioProcessing::
+ kNativeSampleRatesHz[AudioProcessing::kNumNativeSampleRates - 1];
+const int AudioProcessing::kMaxAECMSampleRateHz = kSampleRate16kHz;
+
AudioProcessing* AudioProcessing::Create() {
Config config;
return Create(config, nullptr);
@@ -400,18 +411,16 @@
std::min(api_format_.input_stream().sample_rate_hz(),
api_format_.output_stream().sample_rate_hz());
int fwd_proc_rate;
- if (min_proc_rate > kSampleRate32kHz) {
- fwd_proc_rate = kSampleRate48kHz;
- } else if (min_proc_rate > kSampleRate16kHz) {
- fwd_proc_rate = kSampleRate32kHz;
- } else if (min_proc_rate > kSampleRate8kHz) {
- fwd_proc_rate = kSampleRate16kHz;
- } else {
- fwd_proc_rate = kSampleRate8kHz;
+ for (size_t i = 0; i < kNumNativeSampleRates; ++i) {
+ fwd_proc_rate = kNativeSampleRatesHz[i];
+ if (fwd_proc_rate >= min_proc_rate) {
+ break;
+ }
}
// ...with one exception.
- if (echo_control_mobile_->is_enabled() && min_proc_rate > kSampleRate16kHz) {
- fwd_proc_rate = kSampleRate16kHz;
+ if (echo_control_mobile_->is_enabled() &&
+ min_proc_rate > kMaxAECMSampleRateHz) {
+ fwd_proc_rate = kMaxAECMSampleRateHz;
}
fwd_proc_format_ = StreamConfig(fwd_proc_rate);
@@ -592,7 +601,7 @@
return kBadSampleRateError;
}
if (echo_control_mobile_->is_enabled() &&
- frame->sample_rate_hz_ > kSampleRate16kHz) {
+ frame->sample_rate_hz_ > kMaxAECMSampleRateHz) {
LOG(LS_ERROR) << "AECM only supports 16 or 8 kHz sample rates";
return kUnsupportedComponentError;
}
diff --git a/webrtc/modules/audio_processing/include/audio_processing.h b/webrtc/modules/audio_processing/include/audio_processing.h
index 445d5c8..5eb3b62 100644
--- a/webrtc/modules/audio_processing/include/audio_processing.h
+++ b/webrtc/modules/audio_processing/include/audio_processing.h
@@ -15,6 +15,7 @@
#include <stdio.h> // FILE
#include <vector>
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/platform_file.h"
#include "webrtc/common.h"
#include "webrtc/modules/audio_processing/beamformer/array_util.h"
@@ -128,8 +129,6 @@
bool enabled;
};
-static const int kAudioProcMaxNativeSampleRateHz = 32000;
-
// The Audio Processing Module (APM) provides a collection of voice processing
// components designed for real-time communications software.
//
@@ -471,6 +470,11 @@
kSampleRate48kHz = 48000
};
+ static const int kNativeSampleRatesHz[];
+ static const size_t kNumNativeSampleRates;
+ static const int kMaxNativeSampleRateHz;
+ static const int kMaxAECMSampleRateHz;
+
static const int kChunkSizeMs = 10;
};
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 4ac4b89..fa44785 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -3190,19 +3190,12 @@
CodecInst codec;
GetSendCodec(codec);
- if (!mono_recording_audio_.get()) {
- // Temporary space for DownConvertToCodecFormat.
- mono_recording_audio_.reset(new int16_t[kMaxMonoDataSizeSamples]);
- }
- DownConvertToCodecFormat(audio_data,
- number_of_frames,
- number_of_channels,
- sample_rate,
- codec.channels,
- codec.plfreq,
- mono_recording_audio_.get(),
- &input_resampler_,
- &_audioFrame);
+ // Never upsample or upmix the capture signal here. This should be done at the
+ // end of the send chain.
+ _audioFrame.sample_rate_hz_ = std::min(codec.plfreq, sample_rate);
+ _audioFrame.num_channels_ = std::min(number_of_channels, codec.channels);
+ RemixAndResample(audio_data, number_of_frames, number_of_channels,
+ sample_rate, &input_resampler_, &_audioFrame);
}
uint32_t
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index d9e4575..bf792a5 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -499,7 +499,6 @@
AudioLevel _outputAudioLevel;
bool _externalTransport;
AudioFrame _audioFrame;
- rtc::scoped_ptr<int16_t[]> mono_recording_audio_;
// Downsamples to the codec rate if necessary.
PushResampler<int16_t> input_resampler_;
FilePlayer* _inputFilePlayerPtr;
diff --git a/webrtc/voice_engine/transmit_mixer.cc b/webrtc/voice_engine/transmit_mixer.cc
index 92b9c6e..5e62939 100644
--- a/webrtc/voice_engine/transmit_mixer.cc
+++ b/webrtc/voice_engine/transmit_mixer.cc
@@ -1133,31 +1133,25 @@
int codec_rate;
int num_codec_channels;
GetSendCodecInfo(&codec_rate, &num_codec_channels);
- // TODO(ajm): This currently restricts the sample rate to 32 kHz.
- // See: https://code.google.com/p/webrtc/issues/detail?id=3146
- // When 48 kHz is supported natively by AudioProcessing, this will have
- // to be changed to handle 44.1 kHz.
- int max_sample_rate_hz = kAudioProcMaxNativeSampleRateHz;
- if (audioproc_->echo_control_mobile()->is_enabled()) {
- // AECM only supports 8 and 16 kHz.
- max_sample_rate_hz = 16000;
- }
- codec_rate = std::min(codec_rate, max_sample_rate_hz);
stereo_codec_ = num_codec_channels == 2;
- if (!mono_buffer_.get()) {
- // Temporary space for DownConvertToCodecFormat.
- mono_buffer_.reset(new int16_t[kMaxMonoDataSizeSamples]);
+ // We want to process at the lowest rate possible without losing information.
+ // Choose the lowest native rate at least equal to the input and codec rates.
+ const int min_processing_rate = std::min(sample_rate_hz, codec_rate);
+ for (size_t i = 0; i < AudioProcessing::kNumNativeSampleRates; ++i) {
+ _audioFrame.sample_rate_hz_ = AudioProcessing::kNativeSampleRatesHz[i];
+ if (_audioFrame.sample_rate_hz_ >= min_processing_rate) {
+ break;
+ }
}
- DownConvertToCodecFormat(audio,
- samples_per_channel,
- num_channels,
- sample_rate_hz,
- num_codec_channels,
- codec_rate,
- mono_buffer_.get(),
- &resampler_,
- &_audioFrame);
+ if (audioproc_->echo_control_mobile()->is_enabled()) {
+ // AECM only supports 8 and 16 kHz.
+ _audioFrame.sample_rate_hz_ = std::min(
+ _audioFrame.sample_rate_hz_, AudioProcessing::kMaxAECMSampleRateHz);
+ }
+ _audioFrame.num_channels_ = std::min(num_channels, num_codec_channels);
+ RemixAndResample(audio, samples_per_channel, num_channels, sample_rate_hz,
+ &resampler_, &_audioFrame);
}
int32_t TransmitMixer::RecordAudioToFile(
diff --git a/webrtc/voice_engine/transmit_mixer.h b/webrtc/voice_engine/transmit_mixer.h
index 8bbb421..714efb4 100644
--- a/webrtc/voice_engine/transmit_mixer.h
+++ b/webrtc/voice_engine/transmit_mixer.h
@@ -229,7 +229,6 @@
int32_t _remainingMuteMicTimeMs;
bool stereo_codec_;
bool swap_stereo_channels_;
- rtc::scoped_ptr<int16_t[]> mono_buffer_;
};
} // namespace voe
diff --git a/webrtc/voice_engine/utility.cc b/webrtc/voice_engine/utility.cc
index 82ef076..498620a 100644
--- a/webrtc/voice_engine/utility.cc
+++ b/webrtc/voice_engine/utility.cc
@@ -21,34 +21,43 @@
namespace webrtc {
namespace voe {
-// TODO(ajm): There is significant overlap between RemixAndResample and
-// ConvertToCodecFormat. Consolidate using AudioConverter.
void RemixAndResample(const AudioFrame& src_frame,
PushResampler<int16_t>* resampler,
AudioFrame* dst_frame) {
- const int16_t* audio_ptr = src_frame.data_;
- int audio_ptr_num_channels = src_frame.num_channels_;
+ RemixAndResample(src_frame.data_, src_frame.samples_per_channel_,
+ src_frame.num_channels_, src_frame.sample_rate_hz_,
+ resampler, dst_frame);
+ dst_frame->timestamp_ = src_frame.timestamp_;
+ dst_frame->elapsed_time_ms_ = src_frame.elapsed_time_ms_;
+ dst_frame->ntp_time_ms_ = src_frame.ntp_time_ms_;
+}
+
+void RemixAndResample(const int16_t* src_data,
+ size_t samples_per_channel,
+ int num_channels,
+ int sample_rate_hz,
+ PushResampler<int16_t>* resampler,
+ AudioFrame* dst_frame) {
+ const int16_t* audio_ptr = src_data;
+ int audio_ptr_num_channels = num_channels;
int16_t mono_audio[AudioFrame::kMaxDataSizeSamples];
// Downmix before resampling.
- if (src_frame.num_channels_ == 2 && dst_frame->num_channels_ == 1) {
- AudioFrameOperations::StereoToMono(src_frame.data_,
- src_frame.samples_per_channel_,
+ if (num_channels == 2 && dst_frame->num_channels_ == 1) {
+ AudioFrameOperations::StereoToMono(src_data, samples_per_channel,
mono_audio);
audio_ptr = mono_audio;
audio_ptr_num_channels = 1;
}
- if (resampler->InitializeIfNeeded(src_frame.sample_rate_hz_,
- dst_frame->sample_rate_hz_,
+ if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
audio_ptr_num_channels) == -1) {
- LOG_FERR3(LS_ERROR, InitializeIfNeeded, src_frame.sample_rate_hz_,
+ LOG_FERR3(LS_ERROR, InitializeIfNeeded, sample_rate_hz,
dst_frame->sample_rate_hz_, audio_ptr_num_channels);
assert(false);
}
- const size_t src_length = src_frame.samples_per_channel_ *
- audio_ptr_num_channels;
+ const size_t src_length = samples_per_channel * audio_ptr_num_channels;
int out_length = resampler->Resample(audio_ptr, src_length, dst_frame->data_,
AudioFrame::kMaxDataSizeSamples);
if (out_length == -1) {
@@ -59,66 +68,12 @@
static_cast<size_t>(out_length / audio_ptr_num_channels);
// Upmix after resampling.
- if (src_frame.num_channels_ == 1 && dst_frame->num_channels_ == 2) {
+ if (num_channels == 1 && dst_frame->num_channels_ == 2) {
// The audio in dst_frame really is mono at this point; MonoToStereo will
// set this back to stereo.
dst_frame->num_channels_ = 1;
AudioFrameOperations::MonoToStereo(dst_frame);
}
-
- dst_frame->timestamp_ = src_frame.timestamp_;
- dst_frame->elapsed_time_ms_ = src_frame.elapsed_time_ms_;
- dst_frame->ntp_time_ms_ = src_frame.ntp_time_ms_;
-}
-
-void DownConvertToCodecFormat(const int16_t* src_data,
- size_t samples_per_channel,
- int num_channels,
- int sample_rate_hz,
- int codec_num_channels,
- int codec_rate_hz,
- int16_t* mono_buffer,
- PushResampler<int16_t>* resampler,
- AudioFrame* dst_af) {
- assert(samples_per_channel <= kMaxMonoDataSizeSamples);
- assert(num_channels == 1 || num_channels == 2);
- assert(codec_num_channels == 1 || codec_num_channels == 2);
- dst_af->Reset();
-
- // Never upsample the capture signal here. This should be done at the
- // end of the send chain.
- int destination_rate = std::min(codec_rate_hz, sample_rate_hz);
-
- // If no stereo codecs are in use, we downmix a stereo stream from the
- // device early in the chain, before resampling.
- if (num_channels == 2 && codec_num_channels == 1) {
- AudioFrameOperations::StereoToMono(src_data, samples_per_channel,
- mono_buffer);
- src_data = mono_buffer;
- num_channels = 1;
- }
-
- if (resampler->InitializeIfNeeded(
- sample_rate_hz, destination_rate, num_channels) != 0) {
- LOG_FERR3(LS_ERROR,
- InitializeIfNeeded,
- sample_rate_hz,
- destination_rate,
- num_channels);
- assert(false);
- }
-
- const size_t in_length = samples_per_channel * num_channels;
- int out_length = resampler->Resample(
- src_data, in_length, dst_af->data_, AudioFrame::kMaxDataSizeSamples);
- if (out_length == -1) {
- LOG_FERR3(LS_ERROR, Resample, src_data, in_length, dst_af->data_);
- assert(false);
- }
-
- dst_af->samples_per_channel_ = static_cast<size_t>(out_length / num_channels);
- dst_af->sample_rate_hz_ = destination_rate;
- dst_af->num_channels_ = num_channels;
}
void MixWithSat(int16_t target[],
diff --git a/webrtc/voice_engine/utility.h b/webrtc/voice_engine/utility.h
index 87003c4..cc44533 100644
--- a/webrtc/voice_engine/utility.h
+++ b/webrtc/voice_engine/utility.h
@@ -24,32 +24,26 @@
namespace voe {
-// Upmix or downmix and resample the audio in |src_frame| to |dst_frame|.
-// Expects |dst_frame| to have its sample rate and channels members set to the
-// desired values. Updates the samples per channel member accordingly. No other
-// members will be changed.
+// Upmix or downmix and resample the audio to |dst_frame|. Expects |dst_frame|
+// to have its sample rate and channels members set to the desired values.
+// Updates the |samples_per_channel_| member accordingly.
+//
+// This version has an AudioFrame |src_frame| as input and sets the output
+// |timestamp_|, |elapsed_time_ms_| and |ntp_time_ms_| members equals to the
+// input ones.
void RemixAndResample(const AudioFrame& src_frame,
PushResampler<int16_t>* resampler,
AudioFrame* dst_frame);
-// Downmix and downsample the audio in |src_data| to |dst_af| as necessary,
-// specified by |codec_num_channels| and |codec_rate_hz|. |mono_buffer| is
-// temporary space and must be of sufficient size to hold the downmixed source
-// audio (recommend using a size of kMaxMonoDataSizeSamples).
-//
-// |dst_af| will have its data and format members (sample rate, channels and
-// samples per channel) set appropriately. No other members will be changed.
-// TODO(ajm): For now, this still calls Reset() on |dst_af|. Remove this, as
-// it shouldn't be needed.
-void DownConvertToCodecFormat(const int16_t* src_data,
- size_t samples_per_channel,
- int num_channels,
- int sample_rate_hz,
- int codec_num_channels,
- int codec_rate_hz,
- int16_t* mono_buffer,
- PushResampler<int16_t>* resampler,
- AudioFrame* dst_af);
+// This version has a pointer to the samples |src_data| as input and receives
+// |samples_per_channel|, |num_channels| and |sample_rate_hz| of the data as
+// parameters.
+void RemixAndResample(const int16_t* src_data,
+ size_t samples_per_channel,
+ int num_channels,
+ int sample_rate_hz,
+ PushResampler<int16_t>* resampler,
+ AudioFrame* dst_frame);
void MixWithSat(int16_t target[],
int target_channel,
diff --git a/webrtc/voice_engine/utility_unittest.cc b/webrtc/voice_engine/utility_unittest.cc
index 5f02f51..226e383 100644
--- a/webrtc/voice_engine/utility_unittest.cc
+++ b/webrtc/voice_engine/utility_unittest.cc
@@ -21,11 +21,6 @@
namespace voe {
namespace {
-enum FunctionToTest {
- TestRemixAndResample,
- TestDownConvertToCodecFormat
-};
-
class UtilityTest : public ::testing::Test {
protected:
UtilityTest() {
@@ -36,9 +31,10 @@
golden_frame_.CopyFrom(src_frame_);
}
- void RunResampleTest(int src_channels, int src_sample_rate_hz,
- int dst_channels, int dst_sample_rate_hz,
- FunctionToTest function);
+ void RunResampleTest(int src_channels,
+ int src_sample_rate_hz,
+ int dst_channels,
+ int dst_sample_rate_hz);
PushResampler<int16_t> resampler_;
AudioFrame src_frame_;
@@ -130,8 +126,7 @@
void UtilityTest::RunResampleTest(int src_channels,
int src_sample_rate_hz,
int dst_channels,
- int dst_sample_rate_hz,
- FunctionToTest function) {
+ int dst_sample_rate_hz) {
PushResampler<int16_t> resampler; // Create a new one with every test.
const int16_t kSrcLeft = 30; // Shouldn't overflow for any used sample rate.
const int16_t kSrcRight = 15;
@@ -168,20 +163,7 @@
kInputKernelDelaySamples * dst_channels * 2);
printf("(%d, %d Hz) -> (%d, %d Hz) ", // SNR reported on the same line later.
src_channels, src_sample_rate_hz, dst_channels, dst_sample_rate_hz);
- if (function == TestRemixAndResample) {
- RemixAndResample(src_frame_, &resampler, &dst_frame_);
- } else {
- int16_t mono_buffer[kMaxMonoDataSizeSamples];
- DownConvertToCodecFormat(src_frame_.data_,
- src_frame_.samples_per_channel_,
- src_frame_.num_channels_,
- src_frame_.sample_rate_hz_,
- dst_frame_.num_channels_,
- dst_frame_.sample_rate_hz_,
- mono_buffer,
- &resampler,
- &dst_frame_);
- }
+ RemixAndResample(src_frame_, &resampler, &dst_frame_);
if (src_sample_rate_hz == 96000 && dst_sample_rate_hz == 8000) {
// The sinc resampler gives poor SNR at this extreme conversion, but we
@@ -232,28 +214,7 @@
for (int src_channel = 0; src_channel < kChannelsSize; src_channel++) {
for (int dst_channel = 0; dst_channel < kChannelsSize; dst_channel++) {
RunResampleTest(kChannels[src_channel], kSampleRates[src_rate],
- kChannels[dst_channel], kSampleRates[dst_rate],
- TestRemixAndResample);
- }
- }
- }
- }
-}
-
-TEST_F(UtilityTest, ConvertToCodecFormatSucceeds) {
- const int kSampleRates[] = {8000, 16000, 32000, 44100, 48000, 96000};
- const int kSampleRatesSize = sizeof(kSampleRates) / sizeof(*kSampleRates);
- const int kChannels[] = {1, 2};
- const int kChannelsSize = sizeof(kChannels) / sizeof(*kChannels);
- for (int src_rate = 0; src_rate < kSampleRatesSize; src_rate++) {
- for (int dst_rate = 0; dst_rate < kSampleRatesSize; dst_rate++) {
- for (int src_channel = 0; src_channel < kChannelsSize; src_channel++) {
- for (int dst_channel = 0; dst_channel < kChannelsSize; dst_channel++) {
- if (dst_rate <= src_rate && dst_channel <= src_channel) {
- RunResampleTest(kChannels[src_channel], kSampleRates[src_rate],
- kChannels[src_channel], kSampleRates[dst_rate],
- TestDownConvertToCodecFormat);
- }
+ kChannels[dst_channel], kSampleRates[dst_rate]);
}
}
}
