Implement AudioEncoderPcmU/A classes and convert AudioDecoder tests
BUG=3926
R=kjellander@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7481 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index d6cdf0e..d1f70fa 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -147,7 +147,9 @@
source_set("g711") {
sources = [
+ "codecs/g711/include/audio_encoder_pcm.h",
"codecs/g711/include/g711_interface.h",
+ "codecs/g711/audio_encoder_pcm.cc",
"codecs/g711/g711_interface.c",
"codecs/g711/g711.c",
"codecs/g711/g711.h",
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h
index 1569caf..f8142e2 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -12,7 +12,6 @@
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
#include <algorithm>
-#include <limits>
#include "webrtc/base/checks.h"
#include "webrtc/typedefs.h"
@@ -28,24 +27,27 @@
// Accepts one 10 ms block of input audio (i.e., sample_rate_hz() / 100 *
// num_channels() samples). Multi-channel audio must be sample-interleaved.
// If successful, the encoder produces zero or more bytes of output in
- // |encoded|, and returns the number of bytes. In case of error, -1 is
- // returned. It is an error for the encoder to attempt to produce more than
- // |max_encoded_bytes| bytes of output.
- ssize_t Encode(uint32_t timestamp,
- const int16_t* audio,
- size_t num_samples,
- size_t max_encoded_bytes,
- uint8_t* encoded,
- uint32_t* encoded_timestamp) {
+ // |encoded|, and provides the number of encoded bytes in |encoded_bytes|.
+ // In case of error, false is returned, otherwise true. It is an error for the
+ // encoder to attempt to produce more than |max_encoded_bytes| bytes of
+ // output.
+ bool Encode(uint32_t timestamp,
+ const int16_t* audio,
+ size_t num_samples,
+ size_t max_encoded_bytes,
+ uint8_t* encoded,
+ size_t* encoded_bytes,
+ uint32_t* encoded_timestamp) {
CHECK_EQ(num_samples,
static_cast<size_t>(sample_rate_hz() / 100 * num_channels()));
- ssize_t num_bytes =
- Encode(timestamp, audio, max_encoded_bytes, encoded, encoded_timestamp);
- CHECK_LE(num_bytes,
- static_cast<ssize_t>(std::min(
- max_encoded_bytes,
- static_cast<size_t>(std::numeric_limits<ssize_t>::max()))));
- return num_bytes;
+ bool ret = Encode(timestamp,
+ audio,
+ max_encoded_bytes,
+ encoded,
+ encoded_bytes,
+ encoded_timestamp);
+ CHECK_LE(*encoded_bytes, max_encoded_bytes);
+ return ret;
}
// Returns the input sample rate in Hz, the number of input channels, and the
@@ -56,11 +58,12 @@
virtual int num_10ms_frames_per_packet() const = 0;
protected:
- virtual ssize_t Encode(uint32_t timestamp,
- const int16_t* audio,
- size_t max_encoded_bytes,
- uint8_t* encoded,
- uint32_t* encoded_timestamp) = 0;
+ virtual bool Encode(uint32_t timestamp,
+ const int16_t* audio,
+ size_t max_encoded_bytes,
+ uint8_t* encoded,
+ size_t* encoded_bytes,
+ uint32_t* encoded_timestamp) = 0;
};
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
new file mode 100644
index 0000000..ef22a27
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
@@ -0,0 +1,100 @@
+/*
+ * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h"
+
+#include <limits>
+
+#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
+
+namespace webrtc {
+
+namespace {
+int16_t NumSamplesPerFrame(int num_channels,
+ int frame_size_ms,
+ int sample_rate_hz) {
+ int samples_per_frame = num_channels * frame_size_ms * sample_rate_hz / 1000;
+ CHECK_LE(samples_per_frame, std::numeric_limits<int16_t>::max())
+ << "Frame size too large.";
+ return static_cast<int16_t>(samples_per_frame);
+}
+} // namespace
+
+AudioEncoderPcm::AudioEncoderPcm(const Config& config)
+ : num_channels_(config.num_channels),
+ num_10ms_frames_per_packet_(config.frame_size_ms / 10),
+ full_frame_samples_(NumSamplesPerFrame(num_channels_,
+ config.frame_size_ms,
+ kSampleRateHz)),
+ first_timestamp_in_buffer_(0) {
+ CHECK_EQ(config.frame_size_ms % 10, 0)
+ << "Frame size must be an integer multiple of 10 ms.";
+ speech_buffer_.reserve(full_frame_samples_);
+}
+
+AudioEncoderPcm::~AudioEncoderPcm() {
+}
+
+int AudioEncoderPcm::sample_rate_hz() const {
+ return kSampleRateHz;
+}
+int AudioEncoderPcm::num_channels() const {
+ return num_channels_;
+}
+int AudioEncoderPcm::num_10ms_frames_per_packet() const {
+ return num_10ms_frames_per_packet_;
+}
+
+bool AudioEncoderPcm::Encode(uint32_t timestamp,
+ const int16_t* audio,
+ size_t max_encoded_bytes,
+ uint8_t* encoded,
+ size_t* encoded_bytes,
+ uint32_t* encoded_timestamp) {
+ const int num_samples = sample_rate_hz() / 100 * num_channels();
+ if (speech_buffer_.empty()) {
+ first_timestamp_in_buffer_ = timestamp;
+ }
+ for (int i = 0; i < num_samples; ++i) {
+ speech_buffer_.push_back(audio[i]);
+ }
+ if (speech_buffer_.size() < static_cast<size_t>(full_frame_samples_)) {
+ *encoded_bytes = 0;
+ return true;
+ }
+ CHECK_EQ(speech_buffer_.size(), static_cast<size_t>(full_frame_samples_));
+ int16_t ret = EncodeCall(&speech_buffer_[0], full_frame_samples_, encoded);
+ speech_buffer_.clear();
+ *encoded_timestamp = first_timestamp_in_buffer_;
+ if (ret < 0)
+ return false;
+ *encoded_bytes = static_cast<size_t>(ret);
+ return true;
+}
+
+int16_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
+ size_t input_len,
+ uint8_t* encoded) {
+ return WebRtcG711_EncodeA(NULL,
+ const_cast<int16_t*>(audio),
+ static_cast<int16_t>(input_len),
+ reinterpret_cast<int16_t*>(encoded));
+}
+
+int16_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
+ size_t input_len,
+ uint8_t* encoded) {
+ return WebRtcG711_EncodeU(NULL,
+ const_cast<int16_t*>(audio),
+ static_cast<int16_t>(input_len),
+ reinterpret_cast<int16_t*>(encoded));
+}
+
+} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/codecs/g711/g711.gypi b/webrtc/modules/audio_coding/codecs/g711/g711.gypi
index c39b4af..2b637cf 100644
--- a/webrtc/modules/audio_coding/codecs/g711/g711.gypi
+++ b/webrtc/modules/audio_coding/codecs/g711/g711.gypi
@@ -23,9 +23,11 @@
},
'sources': [
'include/g711_interface.h',
+ 'include/audio_encoder_pcm.h',
'g711_interface.c',
'g711.c',
'g711.h',
+ 'audio_encoder_pcm.cc',
],
},
], # targets
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
new file mode 100644
index 0000000..8133987
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h
@@ -0,0 +1,79 @@
+/*
+ * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_INCLUDE_AUDIO_ENCODER_PCM_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_INCLUDE_AUDIO_ENCODER_PCM_H_
+
+#include <vector>
+
+#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
+
+namespace webrtc {
+
+class AudioEncoderPcm : public AudioEncoder {
+ public:
+ struct Config {
+ Config() : frame_size_ms(20), num_channels(1) {}
+
+ int frame_size_ms;
+ int num_channels;
+ };
+
+ explicit AudioEncoderPcm(const Config& config);
+
+ virtual ~AudioEncoderPcm();
+
+ virtual int sample_rate_hz() const OVERRIDE;
+ virtual int num_channels() const OVERRIDE;
+ virtual int num_10ms_frames_per_packet() const OVERRIDE;
+
+ protected:
+ virtual bool Encode(uint32_t timestamp,
+ const int16_t* audio,
+ size_t max_encoded_bytes,
+ uint8_t* encoded,
+ size_t* encoded_bytes,
+ uint32_t* encoded_timestamp) OVERRIDE;
+
+ virtual int16_t EncodeCall(const int16_t* audio,
+ size_t input_len,
+ uint8_t* encoded) = 0;
+
+ private:
+ static const int kSampleRateHz = 8000;
+ const int num_channels_;
+ const int num_10ms_frames_per_packet_;
+ const int16_t full_frame_samples_;
+ std::vector<int16_t> speech_buffer_;
+ uint32_t first_timestamp_in_buffer_;
+};
+
+class AudioEncoderPcmA : public AudioEncoderPcm {
+ public:
+ explicit AudioEncoderPcmA(const Config& config) : AudioEncoderPcm(config) {}
+
+ protected:
+ virtual int16_t EncodeCall(const int16_t* audio,
+ size_t input_len,
+ uint8_t* encoded) OVERRIDE;
+};
+
+class AudioEncoderPcmU : public AudioEncoderPcm {
+ public:
+ explicit AudioEncoderPcmU(const Config& config) : AudioEncoderPcm(config) {}
+
+ protected:
+ virtual int16_t EncodeCall(const int16_t* audio,
+ size_t input_len,
+ uint8_t* encoded) OVERRIDE;
+};
+
+} // namespace webrtc
+#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_INCLUDE_AUDIO_ENCODER_PCM_H_
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
index 624e6a4..fdb7ac3 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -21,6 +21,7 @@
#include "webrtc/modules/audio_coding/codecs/celt/include/celt_interface.h"
#endif
#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
+#include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h"
#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
@@ -28,6 +29,7 @@
#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
#include "webrtc/system_wrappers/interface/data_log.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/test/testsupport/fileutils.h"
namespace webrtc {
@@ -35,15 +37,16 @@
class AudioDecoderTest : public ::testing::Test {
protected:
AudioDecoderTest()
- : input_fp_(NULL),
- input_(NULL),
- encoded_(NULL),
- decoded_(NULL),
- frame_size_(0),
- data_length_(0),
- encoded_bytes_(0),
- channels_(1),
- decoder_(NULL) {
+ : input_fp_(NULL),
+ input_(NULL),
+ encoded_(NULL),
+ decoded_(NULL),
+ frame_size_(0),
+ data_length_(0),
+ encoded_bytes_(0),
+ channels_(1),
+ output_timestamp_(0),
+ decoder_(NULL) {
input_file_ = webrtc::test::ProjectRootPath() +
"resources/audio_coding/testfile32kHz.pcm";
}
@@ -90,9 +93,25 @@
virtual void InitEncoder() { }
- // This method must be implemented for all tests derived from this class.
- virtual int EncodeFrame(const int16_t* input, size_t input_len,
- uint8_t* output) = 0;
+ // TODO(henrik.lundin) Change return type to size_t once most/all overriding
+ // implementations are gone.
+ virtual int EncodeFrame(const int16_t* input,
+ size_t input_len_samples,
+ uint8_t* output) {
+ size_t enc_len_bytes = 0;
+ for (int i = 0; i < audio_encoder_->num_10ms_frames_per_packet(); ++i) {
+ EXPECT_EQ(0u, enc_len_bytes);
+ EXPECT_TRUE(audio_encoder_->Encode(0,
+ input,
+ audio_encoder_->sample_rate_hz() / 100,
+ data_length_ * 2,
+ output,
+ &enc_len_bytes,
+ &output_timestamp_));
+ }
+ EXPECT_EQ(input_len_samples, enc_len_bytes);
+ return static_cast<int>(enc_len_bytes);
+ }
// Encodes and decodes audio. The absolute difference between the input and
// output is compared vs |tolerance|, and the mean-squared error is compared
@@ -217,7 +236,9 @@
size_t data_length_;
size_t encoded_bytes_;
size_t channels_;
+ uint32_t output_timestamp_;
AudioDecoder* decoder_;
+ scoped_ptr<AudioEncoder> audio_encoder_;
};
class AudioDecoderPcmUTest : public AudioDecoderTest {
@@ -226,17 +247,9 @@
frame_size_ = 160;
data_length_ = 10 * frame_size_;
decoder_ = new AudioDecoderPcmU;
- assert(decoder_);
- }
-
- virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
- uint8_t* output) {
- int enc_len_bytes =
- WebRtcG711_EncodeU(NULL, const_cast<int16_t*>(input),
- static_cast<int>(input_len_samples),
- reinterpret_cast<int16_t*>(output));
- EXPECT_EQ(input_len_samples, static_cast<size_t>(enc_len_bytes));
- return enc_len_bytes;
+ AudioEncoderPcmU::Config config;
+ config.frame_size_ms = static_cast<int>(frame_size_ / 8);
+ audio_encoder_.reset(new AudioEncoderPcmU(config));
}
};
@@ -246,17 +259,9 @@
frame_size_ = 160;
data_length_ = 10 * frame_size_;
decoder_ = new AudioDecoderPcmA;
- assert(decoder_);
- }
-
- virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
- uint8_t* output) {
- int enc_len_bytes =
- WebRtcG711_EncodeA(NULL, const_cast<int16_t*>(input),
- static_cast<int>(input_len_samples),
- reinterpret_cast<int16_t*>(output));
- EXPECT_EQ(input_len_samples, static_cast<size_t>(enc_len_bytes));
- return enc_len_bytes;
+ AudioEncoderPcmA::Config config;
+ config.frame_size_ms = static_cast<int>(frame_size_ / 8);
+ audio_encoder_.reset(new AudioEncoderPcmA(config));
}
};