Make an AudioEncoder subclass for Opus

BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7552 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/codecs/audio_encoder.h

@@ -33,13 +33,13 @@ class AudioEncoder {
   // output.
   bool Encode(uint32_t timestamp,
               const int16_t* audio,
-              size_t num_samples,
+              size_t num_samples_per_channel,
               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()));
+    CHECK_EQ(num_samples_per_channel,
+             static_cast<size_t>(sample_rate_hz() / 100));
     bool ret = Encode(timestamp,
                       audio,
                       max_encoded_bytes,

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc

@@ -0,0 +1,104 @@
+/*
+ *  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/opus/interface/audio_encoder_opus.h"
+
+#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
+
+namespace webrtc {
+
+namespace {
+
+// We always encode at 48 kHz.
+const int kSampleRateHz = 48000;
+
+int DivExact(int a, int b) {
+  CHECK_EQ(a % b, 0);
+  return a / b;
+}
+
+int16_t ClampInt16(size_t x) {
+  return static_cast<int16_t>(
+      std::min(x, static_cast<size_t>(std::numeric_limits<int16_t>::max())));
+}
+
+int16_t CastInt16(size_t x) {
+  DCHECK_LE(x, static_cast<size_t>(std::numeric_limits<int16_t>::max()));
+  return static_cast<int16_t>(x);
+}
+
+}  // namespace
+
+AudioEncoderOpus::Config::Config() : frame_size_ms(20), num_channels(1) {}
+
+bool AudioEncoderOpus::Config::IsOk() const {
+  if (frame_size_ms <= 0 || frame_size_ms % 10 != 0)
+    return false;
+  if (num_channels <= 0)
+    return false;
+  return true;
+}
+
+AudioEncoderOpus::AudioEncoderOpus(const Config& config)
+    : num_10ms_frames_per_packet_(DivExact(config.frame_size_ms, 10)),
+      num_channels_(config.num_channels),
+      samples_per_10ms_frame_(DivExact(kSampleRateHz, 100) * num_channels_) {
+  CHECK(config.IsOk());
+  input_buffer_.reserve(num_10ms_frames_per_packet_ * samples_per_10ms_frame_);
+  CHECK_EQ(0, WebRtcOpus_EncoderCreate(&inst_, num_channels_));
+}
+
+AudioEncoderOpus::~AudioEncoderOpus() {
+  CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
+}
+
+int AudioEncoderOpus::sample_rate_hz() const {
+  return kSampleRateHz;
+}
+
+int AudioEncoderOpus::num_channels() const {
+  return num_channels_;
+}
+
+int AudioEncoderOpus::num_10ms_frames_per_packet() const {
+  return num_10ms_frames_per_packet_;
+}
+
+bool AudioEncoderOpus::Encode(uint32_t timestamp,
+                              const int16_t* audio,
+                              size_t max_encoded_bytes,
+                              uint8_t* encoded,
+                              size_t* encoded_bytes,
+                              uint32_t* encoded_timestamp) {
+  if (input_buffer_.empty())
+    first_timestamp_in_buffer_ = timestamp;
+  input_buffer_.insert(input_buffer_.end(), audio,
+                       audio + samples_per_10ms_frame_);
+  if (input_buffer_.size() < (static_cast<size_t>(num_10ms_frames_per_packet_) *
+                              samples_per_10ms_frame_)) {
+    *encoded_bytes = 0;
+    return true;
+  }
+  CHECK_EQ(input_buffer_.size(),
+           static_cast<size_t>(num_10ms_frames_per_packet_) *
+           samples_per_10ms_frame_);
+  int16_t r = WebRtcOpus_Encode(
+      inst_, &input_buffer_[0],
+      DivExact(CastInt16(input_buffer_.size()), num_channels_),
+      ClampInt16(max_encoded_bytes), encoded);
+  input_buffer_.clear();
+  if (r < 0)
+    return false;
+  *encoded_bytes = r;
+  *encoded_timestamp = first_timestamp_in_buffer_;
+  return true;
+}
+
+}  // namespace webrtc

webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h

@@ -0,0 +1,55 @@
+/*
+ *  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_OPUS_INTERFACE_AUDIO_ENCODER_OPUS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_ENCODER_OPUS_H_
+
+#include <vector>
+
+#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
+
+namespace webrtc {
+
+class AudioEncoderOpus : public AudioEncoder {
+ public:
+  struct Config {
+    Config();
+    bool IsOk() const;
+    int frame_size_ms;
+    int num_channels;
+  };
+
+  explicit AudioEncoderOpus(const Config& config);
+  virtual ~AudioEncoderOpus() OVERRIDE;
+
+  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;
+
+ private:
+  const int num_10ms_frames_per_packet_;
+  const int num_channels_;
+  const int samples_per_10ms_frame_;
+  std::vector<int16_t> input_buffer_;
+  OpusEncInst* inst_;
+  uint32_t first_timestamp_in_buffer_;
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_ENCODER_OPUS_H_

webrtc/modules/audio_coding/codecs/opus/opus.gypi

@@ -27,6 +27,8 @@
         '<(webrtc_root)',
       ],
       'sources': [
+        'audio_encoder_opus.cc',
+        'interface/audio_encoder_opus.h',
         'interface/opus_interface.h',
         'opus_inst.h',
         'opus_interface.c',