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

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 @@ class AudioEncoder {
   // 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 @@ class AudioEncoder {
   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

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

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

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_