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/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