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
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index d1f70fa..547f15f 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -608,6 +608,8 @@
source_set("webrtc_opus") {
sources = [
+ "codecs/opus/audio_encoder_opus.cc",
+ "codecs/opus/interface/audio_encoder_opus.h",
"codecs/opus/interface/opus_interface.h",
"codecs/opus/opus_inst.h",
"codecs/opus/opus_interface.c",
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h
index f8142e2..f9cbe21 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -33,13 +33,13 @@
// 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,
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
new file mode 100644
index 0000000..0a3661f
--- /dev/null
+++ b/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
diff --git a/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h
new file mode 100644
index 0000000..7325b7e
--- /dev/null
+++ b/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_
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus.gypi b/webrtc/modules/audio_coding/codecs/opus/opus.gypi
index 89f0a54..b537285 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus.gypi
+++ b/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',
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
index 3a5a13f..b6c6ba1 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -26,7 +26,7 @@
#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
-#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
+#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
#include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
#include "webrtc/system_wrappers/interface/data_log.h"
@@ -140,17 +140,20 @@
size_t input_len_samples,
uint8_t* output) {
size_t enc_len_bytes = 0;
+ scoped_ptr<int16_t[]> interleaved_input(
+ new int16_t[channels_ * input_len_samples]);
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_));
+
+ // Duplicate the mono input signal to however many channels the test
+ // wants.
+ test::InputAudioFile::DuplicateInterleaved(
+ input, input_len_samples, channels_, interleaved_input.get());
+
+ EXPECT_TRUE(audio_encoder_->Encode(
+ 0, interleaved_input.get(), 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);
}
@@ -636,56 +639,22 @@
frame_size_ = 480;
data_length_ = 10 * frame_size_;
decoder_ = new AudioDecoderOpus(kDecoderOpus);
- assert(decoder_);
- WebRtcOpus_EncoderCreate(&encoder_, 1);
+ AudioEncoderOpus::Config config;
+ config.frame_size_ms = static_cast<int>(frame_size_) / 48;
+ audio_encoder_.reset(new AudioEncoderOpus(config));
}
-
- ~AudioDecoderOpusTest() {
- WebRtcOpus_EncoderFree(encoder_);
- }
-
- virtual void InitEncoder() {}
-
- virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
- uint8_t* output) OVERRIDE {
- int enc_len_bytes = WebRtcOpus_Encode(encoder_, const_cast<int16_t*>(input),
- static_cast<int16_t>(input_len_samples),
- static_cast<int16_t>(data_length_), output);
- EXPECT_GT(enc_len_bytes, 0);
- return enc_len_bytes;
- }
-
- OpusEncInst* encoder_;
};
class AudioDecoderOpusStereoTest : public AudioDecoderOpusTest {
protected:
AudioDecoderOpusStereoTest() : AudioDecoderOpusTest() {
channels_ = 2;
- WebRtcOpus_EncoderFree(encoder_);
delete decoder_;
decoder_ = new AudioDecoderOpus(kDecoderOpus_2ch);
- assert(decoder_);
- WebRtcOpus_EncoderCreate(&encoder_, 2);
- }
-
- virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
- uint8_t* output) OVERRIDE {
- // Create stereo by duplicating each sample in |input|.
- const int input_stereo_samples = static_cast<int>(input_len_samples) * 2;
- scoped_ptr<int16_t[]> input_stereo(new int16_t[input_stereo_samples]);
- test::InputAudioFile::DuplicateInterleaved(
- input, input_len_samples, 2, input_stereo.get());
-
- // Note that the input length is given as samples per channel.
- int enc_len_bytes =
- WebRtcOpus_Encode(encoder_,
- input_stereo.get(),
- static_cast<int16_t>(input_len_samples),
- static_cast<int16_t>(data_length_),
- output);
- EXPECT_GT(enc_len_bytes, 0);
- return enc_len_bytes;
+ AudioEncoderOpus::Config config;
+ config.frame_size_ms = static_cast<int>(frame_size_) / 48;
+ config.num_channels = 2;
+ audio_encoder_.reset(new AudioEncoderOpus(config));
}
};
