Pass audio to AudioEncoder::Encode() in an ArrayView
Instead of in separate pointer and size arguments.
Review URL: https://codereview.webrtc.org/1418423010
Cr-Commit-Position: refs/heads/master@{#10535}
diff --git a/webrtc/base/array_view.h b/webrtc/base/array_view.h
index 019bd8b..02676f1 100644
--- a/webrtc/base/array_view.h
+++ b/webrtc/base/array_view.h
@@ -61,6 +61,7 @@
// is const, because the ArrayView doesn't own the array. (To prevent
// mutation, use ArrayView<const T>.)
size_t size() const { return size_; }
+ bool empty() const { return size_ == 0; }
T* data() const { return data_; }
T& operator[](size_t idx) const {
RTC_DCHECK_LT(idx, size_);
@@ -72,6 +73,15 @@
const T* cbegin() const { return data_; }
const T* cend() const { return data_ + size_; }
+ // Comparing two ArrayViews compares their (pointer,size) pairs; it does
+ // *not* dereference the pointers.
+ friend bool operator==(const ArrayView& a, const ArrayView& b) {
+ return a.data_ == b.data_ && a.size_ == b.size_;
+ }
+ friend bool operator!=(const ArrayView& a, const ArrayView& b) {
+ return !(a == b);
+ }
+
private:
// Invariant: !data_ iff size_ == 0.
void CheckInvariant() const { RTC_DCHECK_EQ(!data_, size_ == 0); }
diff --git a/webrtc/base/array_view_unittest.cc b/webrtc/base/array_view_unittest.cc
index 0d1bff0..8bb1bcc 100644
--- a/webrtc/base/array_view_unittest.cc
+++ b/webrtc/base/array_view_unittest.cc
@@ -214,4 +214,20 @@
}
}
+TEST(ArrayViewTest, TestEmpty) {
+ EXPECT_TRUE(ArrayView<int>().empty());
+ const int a[] = {1, 2, 3};
+ EXPECT_FALSE(ArrayView<const int>(a).empty());
+}
+
+TEST(ArrayViewTest, TestCompare) {
+ int a[] = {1, 2, 3};
+ int b[] = {1, 2, 3};
+ EXPECT_EQ(ArrayView<int>(a), ArrayView<int>(a));
+ EXPECT_EQ(ArrayView<int>(), ArrayView<int>());
+ EXPECT_NE(ArrayView<int>(a), ArrayView<int>(b));
+ EXPECT_NE(ArrayView<int>(a), ArrayView<int>());
+ EXPECT_NE(ArrayView<int>(a), ArrayView<int>(a, 2));
+}
+
} // namespace rtc
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.cc b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
index 6d76300..388b0ff 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.cc
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
@@ -21,13 +21,13 @@
return SampleRateHz();
}
-AudioEncoder::EncodedInfo AudioEncoder::Encode(uint32_t rtp_timestamp,
- const int16_t* audio,
- size_t num_samples_per_channel,
- size_t max_encoded_bytes,
- uint8_t* encoded) {
- RTC_CHECK_EQ(num_samples_per_channel,
- static_cast<size_t>(SampleRateHz() / 100));
+AudioEncoder::EncodedInfo AudioEncoder::Encode(
+ uint32_t rtp_timestamp,
+ rtc::ArrayView<const int16_t> audio,
+ size_t max_encoded_bytes,
+ uint8_t* encoded) {
+ RTC_CHECK_EQ(audio.size(),
+ static_cast<size_t>(NumChannels() * SampleRateHz() / 100));
EncodedInfo info =
EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded);
RTC_CHECK_LE(info.encoded_bytes, max_encoded_bytes);
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h
index cda9d86..553c35e 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -14,6 +14,7 @@
#include <algorithm>
#include <vector>
+#include "webrtc/base/array_view.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -91,13 +92,12 @@
// Encode() checks some preconditions, calls EncodeInternal() which does the
// actual work, and then checks some postconditions.
EncodedInfo Encode(uint32_t rtp_timestamp,
- const int16_t* audio,
- size_t num_samples_per_channel,
+ rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded);
virtual EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
- const int16_t* audio,
+ rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded) = 0;
diff --git a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
index 1215246..e98c537 100644
--- a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
+++ b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
@@ -97,7 +97,7 @@
AudioEncoder::EncodedInfo AudioEncoderCng::EncodeInternal(
uint32_t rtp_timestamp,
- const int16_t* audio,
+ rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded) {
RTC_CHECK_GE(max_encoded_bytes,
@@ -106,9 +106,8 @@
RTC_CHECK_EQ(speech_buffer_.size(),
rtp_timestamps_.size() * samples_per_10ms_frame);
rtp_timestamps_.push_back(rtp_timestamp);
- for (size_t i = 0; i < samples_per_10ms_frame; ++i) {
- speech_buffer_.push_back(audio[i]);
- }
+ RTC_DCHECK_EQ(samples_per_10ms_frame, audio.size());
+ speech_buffer_.insert(speech_buffer_.end(), audio.cbegin(), audio.cend());
const size_t frames_to_encode = speech_encoder_->Num10MsFramesInNextPacket();
if (rtp_timestamps_.size() < frames_to_encode) {
return EncodedInfo();
@@ -242,9 +241,12 @@
const size_t samples_per_10ms_frame = SamplesPer10msFrame();
AudioEncoder::EncodedInfo info;
for (size_t i = 0; i < frames_to_encode; ++i) {
- info = speech_encoder_->Encode(
- rtp_timestamps_.front(), &speech_buffer_[i * samples_per_10ms_frame],
- samples_per_10ms_frame, max_encoded_bytes, encoded);
+ info =
+ speech_encoder_->Encode(rtp_timestamps_.front(),
+ rtc::ArrayView<const int16_t>(
+ &speech_buffer_[i * samples_per_10ms_frame],
+ samples_per_10ms_frame),
+ max_encoded_bytes, encoded);
if (i + 1 == frames_to_encode) {
RTC_CHECK_GT(info.encoded_bytes, 0u) << "Encoder didn't deliver data.";
} else {
diff --git a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
index 0b837a0..ec3f633 100644
--- a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
@@ -75,8 +75,10 @@
void Encode() {
ASSERT_TRUE(cng_) << "Must call CreateCng() first.";
- encoded_info_ = cng_->Encode(timestamp_, audio_, num_audio_samples_10ms_,
- encoded_.size(), &encoded_[0]);
+ encoded_info_ = cng_->Encode(
+ timestamp_,
+ rtc::ArrayView<const int16_t>(audio_, num_audio_samples_10ms_),
+ encoded_.size(), &encoded_[0]);
timestamp_ += static_cast<uint32_t>(num_audio_samples_10ms_);
}
diff --git a/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h b/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h
index 3ca9eb6..c0d61c3 100644
--- a/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h
+++ b/webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h
@@ -57,7 +57,7 @@
size_t Max10MsFramesInAPacket() const override;
int GetTargetBitrate() const override;
EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
- const int16_t* audio,
+ rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded) override;
void Reset() override;
diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
index dde3cc6..6930e2c 100644
--- a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
+++ b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
@@ -88,16 +88,13 @@
AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeInternal(
uint32_t rtp_timestamp,
- const int16_t* audio,
+ rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded) {
- const int num_samples = SampleRateHz() / 100 * NumChannels();
if (speech_buffer_.empty()) {
first_timestamp_in_buffer_ = rtp_timestamp;
}
- for (int i = 0; i < num_samples; ++i) {
- speech_buffer_.push_back(audio[i]);
- }
+ speech_buffer_.insert(speech_buffer_.end(), audio.begin(), audio.end());
if (speech_buffer_.size() < full_frame_samples_) {
return EncodedInfo();
}
diff --git a/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h b/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h
index e532f9b..76eb594 100644
--- a/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h
+++ b/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h
@@ -42,7 +42,7 @@
size_t Max10MsFramesInAPacket() const override;
int GetTargetBitrate() const override;
EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
- const int16_t* audio,
+ rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded) override;
void Reset() override;
diff --git a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
index 43b097f..4c9535e 100644
--- a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
+++ b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
@@ -93,7 +93,7 @@
AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal(
uint32_t rtp_timestamp,
- const int16_t* audio,
+ rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded) {
RTC_CHECK_GE(max_encoded_bytes, MaxEncodedBytes());
diff --git a/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h b/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h
index 12495c5..aad75a1 100644
--- a/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h
+++ b/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h
@@ -42,7 +42,7 @@
size_t Max10MsFramesInAPacket() const override;
int GetTargetBitrate() const override;
EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
- const int16_t* audio,
+ rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded) override;
void Reset() override;
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
index 065dc06..7b497fd 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
+++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
@@ -10,7 +10,7 @@
#include "webrtc/modules/audio_coding/codecs/ilbc/include/audio_encoder_ilbc.h"
-#include <cstring>
+#include <algorithm>
#include <limits>
#include "webrtc/base/checks.h"
#include "webrtc/common_types.h"
@@ -91,7 +91,7 @@
AudioEncoder::EncodedInfo AudioEncoderIlbc::EncodeInternal(
uint32_t rtp_timestamp,
- const int16_t* audio,
+ rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded) {
RTC_DCHECK_GE(max_encoded_bytes, RequiredOutputSizeBytes());
@@ -101,9 +101,9 @@
first_timestamp_in_buffer_ = rtp_timestamp;
// Buffer input.
- std::memcpy(input_buffer_ + kSampleRateHz / 100 * num_10ms_frames_buffered_,
- audio,
- kSampleRateHz / 100 * sizeof(audio[0]));
+ RTC_DCHECK_EQ(static_cast<size_t>(kSampleRateHz / 100), audio.size());
+ std::copy(audio.cbegin(), audio.cend(),
+ input_buffer_ + kSampleRateHz / 100 * num_10ms_frames_buffered_);
// If we don't yet have enough buffered input for a whole packet, we're done
// for now.
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/include/audio_encoder_ilbc.h b/webrtc/modules/audio_coding/codecs/ilbc/include/audio_encoder_ilbc.h
index 2bb3101..e050731 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/include/audio_encoder_ilbc.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/include/audio_encoder_ilbc.h
@@ -41,7 +41,7 @@
size_t Max10MsFramesInAPacket() const override;
int GetTargetBitrate() const override;
EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
- const int16_t* audio,
+ rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded) override;
void Reset() override;
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
index b15ad94..3226877 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
@@ -61,7 +61,7 @@
size_t Max10MsFramesInAPacket() const override;
int GetTargetBitrate() const override;
EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
- const int16_t* audio,
+ rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded) override;
void Reset() override;
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
index 279f80d..4cfd782 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
@@ -115,7 +115,7 @@
template <typename T>
AudioEncoder::EncodedInfo AudioEncoderIsacT<T>::EncodeInternal(
uint32_t rtp_timestamp,
- const int16_t* audio,
+ rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded) {
if (!packet_in_progress_) {
@@ -127,7 +127,7 @@
IsacBandwidthInfo bwinfo = bwinfo_->Get();
T::SetBandwidthInfo(isac_state_, &bwinfo);
}
- int r = T::Encode(isac_state_, audio, encoded);
+ int r = T::Encode(isac_state_, audio.data(), encoded);
RTC_CHECK_GE(r, 0) << "Encode failed (error code "
<< T::GetErrorCode(isac_state_) << ")";
diff --git a/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h b/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h
index 95426d8..29cba8f 100644
--- a/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h
@@ -32,7 +32,7 @@
// Note, we explicitly chose not to create a mock for the Encode method.
MOCK_METHOD4(EncodeInternal,
EncodedInfo(uint32_t timestamp,
- const int16_t* audio,
+ rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded));
MOCK_METHOD0(Reset, void());
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index eac7412..3daf3f9 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -132,13 +132,13 @@
AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeInternal(
uint32_t rtp_timestamp,
- const int16_t* audio,
+ rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded) {
if (input_buffer_.empty())
first_timestamp_in_buffer_ = rtp_timestamp;
- input_buffer_.insert(input_buffer_.end(), audio,
- audio + SamplesPer10msFrame());
+ RTC_DCHECK_EQ(static_cast<size_t>(SamplesPer10msFrame()), audio.size());
+ input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend());
if (input_buffer_.size() <
(static_cast<size_t>(Num10msFramesPerPacket()) * SamplesPer10msFrame())) {
return EncodedInfo();
diff --git a/webrtc/modules/audio_coding/codecs/opus/include/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/include/audio_encoder_opus.h
index 7f2b563..088e2de 100644
--- a/webrtc/modules/audio_coding/codecs/opus/include/audio_encoder_opus.h
+++ b/webrtc/modules/audio_coding/codecs/opus/include/audio_encoder_opus.h
@@ -62,7 +62,7 @@
int GetTargetBitrate() const override;
EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
- const int16_t* audio,
+ rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded) override;
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc b/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
index 4630e44..c059fc5 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
@@ -10,6 +10,7 @@
#include <string>
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h"
#include "webrtc/modules/audio_coding/codecs/opus/opus_inst.h"
#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
@@ -44,8 +45,7 @@
void PrepareSpeechData(int channel, int block_length_ms, int loop_length_ms);
int EncodeDecode(WebRtcOpusEncInst* encoder,
- const int16_t* input_audio,
- size_t input_samples,
+ rtc::ArrayView<const int16_t> input_audio,
WebRtcOpusDecInst* decoder,
int16_t* output_audio,
int16_t* audio_type);
@@ -96,13 +96,14 @@
}
int OpusTest::EncodeDecode(WebRtcOpusEncInst* encoder,
- const int16_t* input_audio,
- size_t input_samples,
+ rtc::ArrayView<const int16_t> input_audio,
WebRtcOpusDecInst* decoder,
int16_t* output_audio,
int16_t* audio_type) {
- int encoded_bytes_int = WebRtcOpus_Encode(encoder, input_audio, input_samples,
- kMaxBytes, bitstream_);
+ int encoded_bytes_int = WebRtcOpus_Encode(
+ encoder, input_audio.data(),
+ rtc::CheckedDivExact(input_audio.size(), static_cast<size_t>(channels_)),
+ kMaxBytes, bitstream_);
EXPECT_GE(encoded_bytes_int, 0);
encoded_bytes_ = static_cast<size_t>(encoded_bytes_int);
int est_len = WebRtcOpus_DurationEst(decoder, bitstream_, encoded_bytes_);
@@ -129,8 +130,7 @@
channels_ == 1 ? 32000 : 64000));
// Set input audio as silence.
- int16_t* silence = new int16_t[kOpus20msFrameSamples * channels_];
- memset(silence, 0, sizeof(int16_t) * kOpus20msFrameSamples * channels_);
+ std::vector<int16_t> silence(kOpus20msFrameSamples * channels_, 0);
// Setting DTX.
EXPECT_EQ(0, dtx ? WebRtcOpus_EnableDtx(opus_encoder_) :
@@ -142,9 +142,8 @@
for (int i = 0; i < 100; ++i) {
EXPECT_EQ(kOpus20msFrameSamples,
static_cast<size_t>(EncodeDecode(
- opus_encoder_, speech_data_.GetNextBlock(),
- kOpus20msFrameSamples, opus_decoder_, output_data_decode,
- &audio_type)));
+ opus_encoder_, speech_data_.GetNextBlock(), opus_decoder_,
+ output_data_decode, &audio_type)));
// If not DTX, it should never enter DTX mode. If DTX, we do not care since
// whether it enters DTX depends on the signal type.
if (!dtx) {
@@ -158,10 +157,9 @@
// We input some silent segments. In DTX mode, the encoder will stop sending.
// However, DTX may happen after a while.
for (int i = 0; i < 30; ++i) {
- EXPECT_EQ(kOpus20msFrameSamples,
- static_cast<size_t>(EncodeDecode(
- opus_encoder_, silence, kOpus20msFrameSamples, opus_decoder_,
- output_data_decode, &audio_type)));
+ EXPECT_EQ(kOpus20msFrameSamples, static_cast<size_t>(EncodeDecode(
+ opus_encoder_, silence, opus_decoder_,
+ output_data_decode, &audio_type)));
if (!dtx) {
EXPECT_GT(encoded_bytes_, 1U);
EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
@@ -183,9 +181,9 @@
// DTX mode is maintained 19 frames.
for (int i = 0; i < 19; ++i) {
EXPECT_EQ(kOpus20msFrameSamples,
- static_cast<size_t>(EncodeDecode(
- opus_encoder_, silence, kOpus20msFrameSamples,
- opus_decoder_, output_data_decode, &audio_type)));
+ static_cast<size_t>(
+ EncodeDecode(opus_encoder_, silence, opus_decoder_,
+ output_data_decode, &audio_type)));
if (dtx) {
EXPECT_EQ(0U, encoded_bytes_) // Send 0 byte.
<< "Opus should have entered DTX mode.";
@@ -201,10 +199,9 @@
}
// Quit DTX after 19 frames.
- EXPECT_EQ(kOpus20msFrameSamples,
- static_cast<size_t>(EncodeDecode(
- opus_encoder_, silence, kOpus20msFrameSamples, opus_decoder_,
- output_data_decode, &audio_type)));
+ EXPECT_EQ(kOpus20msFrameSamples, static_cast<size_t>(EncodeDecode(
+ opus_encoder_, silence, opus_decoder_,
+ output_data_decode, &audio_type)));
EXPECT_GT(encoded_bytes_, 1U);
EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
@@ -212,10 +209,9 @@
EXPECT_EQ(0, audio_type); // Speech.
// Enters DTX again immediately.
- EXPECT_EQ(kOpus20msFrameSamples,
- static_cast<size_t>(EncodeDecode(
- opus_encoder_, silence, kOpus20msFrameSamples, opus_decoder_,
- output_data_decode, &audio_type)));
+ EXPECT_EQ(kOpus20msFrameSamples, static_cast<size_t>(EncodeDecode(
+ opus_encoder_, silence, opus_decoder_,
+ output_data_decode, &audio_type)));
if (dtx) {
EXPECT_EQ(1U, encoded_bytes_); // Send 1 byte.
EXPECT_EQ(1, opus_encoder_->in_dtx_mode);
@@ -232,10 +228,9 @@
silence[0] = 10000;
if (dtx) {
// Verify that encoder/decoder can jump out from DTX mode.
- EXPECT_EQ(kOpus20msFrameSamples,
- static_cast<size_t>(EncodeDecode(
- opus_encoder_, silence, kOpus20msFrameSamples, opus_decoder_,
- output_data_decode, &audio_type)));
+ EXPECT_EQ(kOpus20msFrameSamples, static_cast<size_t>(EncodeDecode(
+ opus_encoder_, silence, opus_decoder_,
+ output_data_decode, &audio_type)));
EXPECT_GT(encoded_bytes_, 1U);
EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
@@ -244,7 +239,6 @@
// Free memory.
delete[] output_data_decode;
- delete[] silence;
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
}
@@ -314,10 +308,9 @@
int16_t audio_type;
int16_t* output_data_decode = new int16_t[kOpus20msFrameSamples * channels_];
EXPECT_EQ(kOpus20msFrameSamples,
- static_cast<size_t>(EncodeDecode(
- opus_encoder_, speech_data_.GetNextBlock(),
- kOpus20msFrameSamples, opus_decoder_, output_data_decode,
- &audio_type)));
+ static_cast<size_t>(
+ EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+ opus_decoder_, output_data_decode, &audio_type)));
// Free memory.
delete[] output_data_decode;
@@ -374,10 +367,9 @@
int16_t audio_type;
int16_t* output_data_decode = new int16_t[kOpus20msFrameSamples * channels_];
EXPECT_EQ(kOpus20msFrameSamples,
- static_cast<size_t>(EncodeDecode(
- opus_encoder_, speech_data_.GetNextBlock(),
- kOpus20msFrameSamples, opus_decoder_, output_data_decode,
- &audio_type)));
+ static_cast<size_t>(
+ EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+ opus_decoder_, output_data_decode, &audio_type)));
WebRtcOpus_DecoderInit(opus_decoder_);
@@ -513,10 +505,9 @@
int16_t audio_type;
int16_t* output_data_decode = new int16_t[kOpus20msFrameSamples * channels_];
EXPECT_EQ(kOpus20msFrameSamples,
- static_cast<size_t>(EncodeDecode(
- opus_encoder_, speech_data_.GetNextBlock(),
- kOpus20msFrameSamples, opus_decoder_, output_data_decode,
- &audio_type)));
+ static_cast<size_t>(
+ EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+ opus_decoder_, output_data_decode, &audio_type)));
// Call decoder PLC.
int16_t* plc_buffer = new int16_t[kOpus20msFrameSamples * channels_];
@@ -542,10 +533,12 @@
EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
// 10 ms. We use only first 10 ms of a 20 ms block.
- int encoded_bytes_int = WebRtcOpus_Encode(opus_encoder_,
- speech_data_.GetNextBlock(),
- kOpus10msFrameSamples,
- kMaxBytes, bitstream_);
+ auto speech_block = speech_data_.GetNextBlock();
+ int encoded_bytes_int = WebRtcOpus_Encode(
+ opus_encoder_, speech_block.data(),
+ rtc::CheckedDivExact(speech_block.size(),
+ 2 * static_cast<size_t>(channels_)),
+ kMaxBytes, bitstream_);
EXPECT_GE(encoded_bytes_int, 0);
EXPECT_EQ(kOpus10msFrameSamples,
static_cast<size_t>(WebRtcOpus_DurationEst(
@@ -553,10 +546,11 @@
static_cast<size_t>(encoded_bytes_int))));
// 20 ms
- encoded_bytes_int = WebRtcOpus_Encode(opus_encoder_,
- speech_data_.GetNextBlock(),
- kOpus20msFrameSamples,
- kMaxBytes, bitstream_);
+ speech_block = speech_data_.GetNextBlock();
+ encoded_bytes_int = WebRtcOpus_Encode(
+ opus_encoder_, speech_block.data(),
+ rtc::CheckedDivExact(speech_block.size(), static_cast<size_t>(channels_)),
+ kMaxBytes, bitstream_);
EXPECT_GE(encoded_bytes_int, 0);
EXPECT_EQ(kOpus20msFrameSamples,
static_cast<size_t>(WebRtcOpus_DurationEst(
@@ -594,10 +588,12 @@
OpusRepacketizer* rp = opus_repacketizer_create();
for (int idx = 0; idx < kPackets; idx++) {
- encoded_bytes_ = WebRtcOpus_Encode(opus_encoder_,
- speech_data_.GetNextBlock(),
- kOpus20msFrameSamples, kMaxBytes,
- bitstream_);
+ auto speech_block = speech_data_.GetNextBlock();
+ encoded_bytes_ =
+ WebRtcOpus_Encode(opus_encoder_, speech_block.data(),
+ rtc::CheckedDivExact(speech_block.size(),
+ static_cast<size_t>(channels_)),
+ kMaxBytes, bitstream_);
EXPECT_EQ(OPUS_OK, opus_repacketizer_cat(rp, bitstream_, encoded_bytes_));
}
diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
index a19d194..177c19a 100644
--- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
+++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
@@ -54,12 +54,11 @@
AudioEncoder::EncodedInfo AudioEncoderCopyRed::EncodeInternal(
uint32_t rtp_timestamp,
- const int16_t* audio,
+ rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded) {
- EncodedInfo info = speech_encoder_->Encode(
- rtp_timestamp, audio, static_cast<size_t>(SampleRateHz() / 100),
- max_encoded_bytes, encoded);
+ EncodedInfo info =
+ speech_encoder_->Encode(rtp_timestamp, audio, max_encoded_bytes, encoded);
RTC_CHECK_GE(max_encoded_bytes,
info.encoded_bytes + secondary_info_.encoded_bytes);
RTC_CHECK(info.redundant.empty()) << "Cannot use nested redundant encoders.";
diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
index 7837010..d7d3a66 100644
--- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
+++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
@@ -44,7 +44,7 @@
size_t Max10MsFramesInAPacket() const override;
int GetTargetBitrate() const override;
EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
- const int16_t* audio,
+ rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded) override;
void Reset() override;
diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
index cb50652..c4c3910 100644
--- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
@@ -60,8 +60,10 @@
void Encode() {
ASSERT_TRUE(red_.get() != NULL);
- encoded_info_ = red_->Encode(timestamp_, audio_, num_audio_samples_10ms,
- encoded_.size(), &encoded_[0]);
+ encoded_info_ = red_->Encode(
+ timestamp_,
+ rtc::ArrayView<const int16_t>(audio_, num_audio_samples_10ms),
+ encoded_.size(), &encoded_[0]);
timestamp_ += num_audio_samples_10ms;
}
@@ -83,7 +85,7 @@
}
AudioEncoder::EncodedInfo Encode(uint32_t timestamp,
- const int16_t* audio,
+ rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded) {
if (write_payload_) {
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index 260f8a8..3b8b140 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -149,7 +149,9 @@
encode_buffer_.SetSize(audio_encoder->MaxEncodedBytes());
encoded_info = audio_encoder->Encode(
- rtp_timestamp, input_data.audio, input_data.length_per_channel,
+ rtp_timestamp, rtc::ArrayView<const int16_t>(
+ input_data.audio, input_data.audio_channel *
+ input_data.length_per_channel),
encode_buffer_.size(), encode_buffer_.data());
encode_buffer_.SetSize(encoded_info.encoded_bytes);
bitrate_logger_.MaybeLog(audio_encoder->GetTargetBitrate() / 1000);
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
index 3aee344..cfceb0d 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
@@ -656,7 +656,11 @@
}
void InsertAudio() {
- memcpy(input_frame_.data_, audio_loop_.GetNextBlock(), kNumSamples10ms);
+ // TODO(kwiberg): Use std::copy here. Might be complications because AFAICS
+ // this call confuses the number of samples with the number of bytes, and
+ // ends up copying only half of what it should.
+ memcpy(input_frame_.data_, audio_loop_.GetNextBlock().data(),
+ kNumSamples10ms);
AudioCodingModuleTestOldApi::InsertAudio();
}
@@ -774,9 +778,9 @@
// Encode new frame.
uint32_t input_timestamp = rtp_header_.header.timestamp;
while (info.encoded_bytes == 0) {
- info = isac_encoder_->Encode(
- input_timestamp, audio_loop_.GetNextBlock(), kNumSamples10ms,
- max_encoded_bytes, encoded.get());
+ info =
+ isac_encoder_->Encode(input_timestamp, audio_loop_.GetNextBlock(),
+ max_encoded_bytes, encoded.get());
input_timestamp += 160; // 10 ms at 16 kHz.
}
EXPECT_EQ(rtp_header_.header.timestamp + kPacketSizeSamples,
diff --git a/webrtc/modules/audio_coding/main/acm2/codec_owner_unittest.cc b/webrtc/modules/audio_coding/main/acm2/codec_owner_unittest.cc
index 6c23261..6c4d38f 100644
--- a/webrtc/modules/audio_coding/main/acm2/codec_owner_unittest.cc
+++ b/webrtc/modules/audio_coding/main/acm2/codec_owner_unittest.cc
@@ -46,8 +46,8 @@
int expected_send_even_if_empty) {
uint8_t out[kPacketSizeSamples];
AudioEncoder::EncodedInfo encoded_info;
- encoded_info = codec_owner_.Encoder()->Encode(
- timestamp_, kZeroData, kDataLengthSamples, kPacketSizeSamples, out);
+ encoded_info = codec_owner_.Encoder()->Encode(timestamp_, kZeroData,
+ kPacketSizeSamples, out);
timestamp_ += kDataLengthSamples;
EXPECT_TRUE(encoded_info.redundant.empty());
EXPECT_EQ(expected_out_length, encoded_info.encoded_bytes);
@@ -146,24 +146,26 @@
AudioEncoder::EncodedInfo info;
EXPECT_CALL(external_encoder, SampleRateHz())
.WillRepeatedly(Return(kSampleRateHz));
+ EXPECT_CALL(external_encoder, NumChannels()).WillRepeatedly(Return(1));
{
InSequence s;
info.encoded_timestamp = 0;
EXPECT_CALL(external_encoder,
- EncodeInternal(0, audio, arraysize(encoded), encoded))
+ EncodeInternal(0, rtc::ArrayView<const int16_t>(audio),
+ arraysize(encoded), encoded))
.WillOnce(Return(info));
EXPECT_CALL(external_encoder, Mark("A"));
EXPECT_CALL(external_encoder, Mark("B"));
info.encoded_timestamp = 2;
EXPECT_CALL(external_encoder,
- EncodeInternal(2, audio, arraysize(encoded), encoded))
+ EncodeInternal(2, rtc::ArrayView<const int16_t>(audio),
+ arraysize(encoded), encoded))
.WillOnce(Return(info));
EXPECT_CALL(external_encoder, Die());
}
- info = codec_owner_.Encoder()->Encode(0, audio, arraysize(audio),
- arraysize(encoded), encoded);
+ info = codec_owner_.Encoder()->Encode(0, audio, arraysize(encoded), encoded);
EXPECT_EQ(0u, info.encoded_timestamp);
external_encoder.Mark("A");
@@ -172,14 +174,12 @@
codec_inst.pacsize = kPacketSizeSamples;
ASSERT_TRUE(codec_owner_.SetEncoders(codec_inst, -1, VADNormal, -1));
// Don't expect any more calls to the external encoder.
- info = codec_owner_.Encoder()->Encode(1, audio, arraysize(audio),
- arraysize(encoded), encoded);
+ info = codec_owner_.Encoder()->Encode(1, audio, arraysize(encoded), encoded);
external_encoder.Mark("B");
// Change back to external encoder again.
codec_owner_.SetEncoders(&external_encoder, -1, VADNormal, -1);
- info = codec_owner_.Encoder()->Encode(2, audio, arraysize(audio),
- arraysize(encoded), encoded);
+ info = codec_owner_.Encoder()->Encode(2, audio, arraysize(encoded), encoded);
EXPECT_EQ(2u, info.encoded_timestamp);
}
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
index 8f82fb1..accae85 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -158,7 +158,10 @@
interleaved_input.get());
encoded_info_ = audio_encoder_->Encode(
- 0, interleaved_input.get(), audio_encoder_->SampleRateHz() / 100,
+ 0, rtc::ArrayView<const int16_t>(interleaved_input.get(),
+ audio_encoder_->NumChannels() *
+ audio_encoder_->SampleRateHz() /
+ 100),
data_length_ * 2, output);
}
EXPECT_EQ(payload_type_, encoded_info_.payload_type);
diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
index 4340f54..6f47dd1 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -939,8 +939,10 @@
uint32_t receive_timestamp = 0;
for (int n = 0; n < 10; ++n) { // Insert few packets and get audio.
- size_t enc_len_bytes = WebRtcPcm16b_Encode(
- input.GetNextBlock(), expected_samples_per_channel, payload);
+ auto block = input.GetNextBlock();
+ ASSERT_EQ(expected_samples_per_channel, block.size());
+ size_t enc_len_bytes =
+ WebRtcPcm16b_Encode(block.data(), block.size(), payload);
ASSERT_EQ(enc_len_bytes, expected_samples_per_channel * 2);
number_channels = 0;
diff --git a/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc b/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
index 2042e0d..b61bfde 100644
--- a/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
+++ b/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
@@ -66,8 +66,10 @@
uint32_t dummy_timestamp = 0;
AudioEncoder::EncodedInfo info;
do {
- info = encoder_->Encode(dummy_timestamp, &in_data[encoded_samples],
- kFrameSizeSamples, max_bytes, payload);
+ info = encoder_->Encode(dummy_timestamp,
+ rtc::ArrayView<const int16_t>(
+ in_data + encoded_samples, kFrameSizeSamples),
+ max_bytes, payload);
encoded_samples += kFrameSizeSamples;
} while (info.encoded_bytes == 0);
return rtc::checked_cast<int>(info.encoded_bytes);
diff --git a/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc b/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
index 422a9fa..01c3964 100644
--- a/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
+++ b/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
@@ -66,8 +66,10 @@
uint32_t dummy_timestamp = 0;
AudioEncoder::EncodedInfo info;
do {
- info = encoder_->Encode(dummy_timestamp, &in_data[encoded_samples],
- kFrameSizeSamples, max_bytes, payload);
+ info = encoder_->Encode(dummy_timestamp,
+ rtc::ArrayView<const int16_t>(
+ in_data + encoded_samples, kFrameSizeSamples),
+ max_bytes, payload);
encoded_samples += kFrameSizeSamples;
} while (info.encoded_bytes == 0);
return rtc::checked_cast<int>(info.encoded_bytes);
diff --git a/webrtc/modules/audio_coding/neteq/tools/audio_loop.cc b/webrtc/modules/audio_coding/neteq/tools/audio_loop.cc
index 2d2a7e3..eed9575 100644
--- a/webrtc/modules/audio_coding/neteq/tools/audio_loop.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/audio_loop.cc
@@ -43,13 +43,14 @@
return true;
}
-const int16_t* AudioLoop::GetNextBlock() {
+rtc::ArrayView<const int16_t> AudioLoop::GetNextBlock() {
// Check that the AudioLoop is initialized.
- if (block_length_samples_ == 0) return NULL;
+ if (block_length_samples_ == 0)
+ return rtc::ArrayView<const int16_t>();
const int16_t* output_ptr = &audio_array_[next_index_];
next_index_ = (next_index_ + block_length_samples_) % loop_length_samples_;
- return output_ptr;
+ return rtc::ArrayView<const int16_t>(output_ptr, block_length_samples_);
}
diff --git a/webrtc/modules/audio_coding/neteq/tools/audio_loop.h b/webrtc/modules/audio_coding/neteq/tools/audio_loop.h
index a897ee5..14e20f6 100644
--- a/webrtc/modules/audio_coding/neteq/tools/audio_loop.h
+++ b/webrtc/modules/audio_coding/neteq/tools/audio_loop.h
@@ -13,6 +13,7 @@
#include <string>
+#include "webrtc/base/array_view.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/typedefs.h"
@@ -40,10 +41,9 @@
bool Init(const std::string file_name, size_t max_loop_length_samples,
size_t block_length_samples);
- // Returns a pointer to the next block of audio. The number given as
- // |block_length_samples| to the Init() function determines how many samples
- // that can be safely read from the pointer.
- const int16_t* GetNextBlock();
+ // Returns a (pointer,size) pair for the next block of audio. The size is
+ // equal to the |block_length_samples| Init() argument.
+ rtc::ArrayView<const int16_t> GetNextBlock();
private:
size_t next_index_;
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
index 9fe4dff..dbea1c6 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
@@ -62,11 +62,12 @@
bool drift_flipped = false;
int32_t packet_input_time_ms =
rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header);
- const int16_t* input_samples = audio_loop.GetNextBlock();
- if (!input_samples) exit(1);
+ auto input_samples = audio_loop.GetNextBlock();
+ if (input_samples.empty())
+ exit(1);
uint8_t input_payload[kInputBlockSizeSamples * sizeof(int16_t)];
- size_t payload_len =
- WebRtcPcm16b_Encode(input_samples, kInputBlockSizeSamples, input_payload);
+ size_t payload_len = WebRtcPcm16b_Encode(input_samples.data(),
+ input_samples.size(), input_payload);
assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t));
// Main loop.
@@ -93,10 +94,10 @@
kInputBlockSizeSamples,
&rtp_header);
input_samples = audio_loop.GetNextBlock();
- if (!input_samples) return -1;
- payload_len = WebRtcPcm16b_Encode(const_cast<int16_t*>(input_samples),
- kInputBlockSizeSamples,
- input_payload);
+ if (input_samples.empty())
+ return -1;
+ payload_len = WebRtcPcm16b_Encode(input_samples.data(),
+ input_samples.size(), input_payload);
assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t));
}
