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));
   }
 };