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-rw-r--r--webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc4
-rw-r--r--webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h (renamed from webrtc/modules/audio_coding/codecs/g711/include/audio_decoder_pcm.h)6
-rw-r--r--webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc34
-rw-r--r--webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h (renamed from webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h)20
-rw-r--r--webrtc/modules/audio_coding/codecs/g711/g711.gypi20
-rw-r--r--webrtc/modules/audio_coding/codecs/g711/g711_interface.h (renamed from webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h)6
-rw-r--r--webrtc/modules/audio_coding/codecs/g711/test/testG711.cc2
7 files changed, 35 insertions, 57 deletions
diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc b/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
index 12306d9167..9757b4a010 100644
--- a/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
+++ b/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
@@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/g711/include/audio_decoder_pcm.h"
+#include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
-#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
+#include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_coding/codecs/g711/include/audio_decoder_pcm.h b/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
index 7bc37d3b7a..9dc3a6fd7a 100644
--- a/webrtc/modules/audio_coding/codecs/g711/include/audio_decoder_pcm.h
+++ b/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_INCLUDE_AUDIO_DECODER_PCM_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_INCLUDE_AUDIO_DECODER_PCM_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
@@ -60,4 +60,4 @@ class AudioDecoderPcmA final : public AudioDecoder {
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_INCLUDE_AUDIO_DECODER_PCM_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
index dde3cc6799..ff61db8e8d 100644
--- a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
+++ b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
@@ -8,27 +8,18 @@
* 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 "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
#include <limits>
#include "webrtc/base/checks.h"
#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
+#include "webrtc/modules/audio_coding/codecs/g711/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;
- RTC_CHECK_LE(samples_per_frame, std::numeric_limits<int16_t>::max())
- << "Frame size too large.";
- return static_cast<int16_t>(samples_per_frame);
-}
-
template <typename T>
typename T::Config CreateConfig(const CodecInst& codec_inst) {
typename T::Config config;
@@ -50,9 +41,8 @@ AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz)
payload_type_(config.payload_type),
num_10ms_frames_per_packet_(
static_cast<size_t>(config.frame_size_ms / 10)),
- full_frame_samples_(NumSamplesPerFrame(config.num_channels,
- config.frame_size_ms,
- sample_rate_hz_)),
+ full_frame_samples_(
+ config.num_channels * config.frame_size_ms * sample_rate_hz / 1000),
first_timestamp_in_buffer_(0) {
RTC_CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz";
RTC_CHECK_EQ(config.frame_size_ms % 10, 0)
@@ -70,7 +60,7 @@ int AudioEncoderPcm::SampleRateHz() const {
return sample_rate_hz_;
}
-int AudioEncoderPcm::NumChannels() const {
+size_t AudioEncoderPcm::NumChannels() const {
return num_channels_;
}
@@ -83,21 +73,19 @@ size_t AudioEncoderPcm::Max10MsFramesInAPacket() const {
}
int AudioEncoderPcm::GetTargetBitrate() const {
- return 8 * BytesPerSample() * SampleRateHz() * NumChannels();
+ return static_cast<int>(
+ 8 * BytesPerSample() * SampleRateHz() * NumChannels());
}
AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeInternal(
uint32_t rtp_timestamp,
- const int16_t* audio,
+ rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded) {
- const int num_samples = SampleRateHz() / 100 * NumChannels();
if (speech_buffer_.empty()) {
first_timestamp_in_buffer_ = rtp_timestamp;
}
- for (int i = 0; i < num_samples; ++i) {
- speech_buffer_.push_back(audio[i]);
- }
+ speech_buffer_.insert(speech_buffer_.end(), audio.begin(), audio.end());
if (speech_buffer_.size() < full_frame_samples_) {
return EncodedInfo();
}
@@ -125,7 +113,7 @@ size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
return WebRtcG711_EncodeA(audio, input_len, encoded);
}
-int AudioEncoderPcmA::BytesPerSample() const {
+size_t AudioEncoderPcmA::BytesPerSample() const {
return 1;
}
@@ -138,7 +126,7 @@ size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
return WebRtcG711_EncodeU(audio, input_len, encoded);
}
-int AudioEncoderPcmU::BytesPerSample() const {
+size_t AudioEncoderPcmU::BytesPerSample() const {
return 1;
}
diff --git a/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
index e532f9b1bc..b839488628 100644
--- a/webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h
+++ b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
@@ -8,8 +8,8 @@
* 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_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
#include <vector>
@@ -25,7 +25,7 @@ class AudioEncoderPcm : public AudioEncoder {
bool IsOk() const;
int frame_size_ms;
- int num_channels;
+ size_t num_channels;
int payload_type;
protected:
@@ -37,12 +37,12 @@ class AudioEncoderPcm : public AudioEncoder {
size_t MaxEncodedBytes() const override;
int SampleRateHz() const override;
- int NumChannels() const override;
+ size_t NumChannels() const override;
size_t Num10MsFramesInNextPacket() const override;
size_t Max10MsFramesInAPacket() const override;
int GetTargetBitrate() const override;
EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
- const int16_t* audio,
+ rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded) override;
void Reset() override;
@@ -54,11 +54,11 @@ class AudioEncoderPcm : public AudioEncoder {
size_t input_len,
uint8_t* encoded) = 0;
- virtual int BytesPerSample() const = 0;
+ virtual size_t BytesPerSample() const = 0;
private:
const int sample_rate_hz_;
- const int num_channels_;
+ const size_t num_channels_;
const int payload_type_;
const size_t num_10ms_frames_per_packet_;
const size_t full_frame_samples_;
@@ -83,7 +83,7 @@ class AudioEncoderPcmA final : public AudioEncoderPcm {
size_t input_len,
uint8_t* encoded) override;
- int BytesPerSample() const override;
+ size_t BytesPerSample() const override;
private:
static const int kSampleRateHz = 8000;
@@ -105,7 +105,7 @@ class AudioEncoderPcmU final : public AudioEncoderPcm {
size_t input_len,
uint8_t* encoded) override;
- int BytesPerSample() const override;
+ size_t BytesPerSample() const override;
private:
static const int kSampleRateHz = 8000;
@@ -114,4 +114,4 @@ class AudioEncoderPcmU final : public AudioEncoderPcm {
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_INCLUDE_AUDIO_ENCODER_PCM_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
diff --git a/webrtc/modules/audio_coding/codecs/g711/g711.gypi b/webrtc/modules/audio_coding/codecs/g711/g711.gypi
index d35d7874e7..4b902809ea 100644
--- a/webrtc/modules/audio_coding/codecs/g711/g711.gypi
+++ b/webrtc/modules/audio_coding/codecs/g711/g711.gypi
@@ -14,25 +14,15 @@
'dependencies': [
'audio_encoder_interface',
],
- 'include_dirs': [
- 'include',
- '<(webrtc_root)',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- 'include',
- '<(webrtc_root)',
- ],
- },
'sources': [
- 'include/g711_interface.h',
- 'include/audio_decoder_pcm.h',
- 'include/audio_encoder_pcm.h',
+ 'audio_decoder_pcm.cc',
+ 'audio_decoder_pcm.h',
+ 'audio_encoder_pcm.cc',
+ 'audio_encoder_pcm.h',
'g711_interface.c',
+ 'g711_interface.h',
'g711.c',
'g711.h',
- 'audio_decoder_pcm.cc',
- 'audio_encoder_pcm.cc',
],
},
], # targets
diff --git a/webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h b/webrtc/modules/audio_coding/codecs/g711/g711_interface.h
index f9867f4504..00854bbb2c 100644
--- a/webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h
+++ b/webrtc/modules/audio_coding/codecs/g711/g711_interface.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef MODULES_AUDIO_CODING_CODECS_G711_MAIN_INCLUDE_G711_INTERFACE_H_
-#define MODULES_AUDIO_CODING_CODECS_G711_MAIN_INCLUDE_G711_INTERFACE_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
#include "webrtc/typedefs.h"
@@ -132,4 +132,4 @@ int16_t WebRtcG711_Version(char* version, int16_t lenBytes);
}
#endif
-#endif /* MODULES_AUDIO_CODING_CODECS_G711_MAIN_INCLUDE_G711_INCLUDE_H_ */
+#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
diff --git a/webrtc/modules/audio_coding/codecs/g711/test/testG711.cc b/webrtc/modules/audio_coding/codecs/g711/test/testG711.cc
index 94248f7a66..5675b1f8b0 100644
--- a/webrtc/modules/audio_coding/codecs/g711/test/testG711.cc
+++ b/webrtc/modules/audio_coding/codecs/g711/test/testG711.cc
@@ -17,7 +17,7 @@
#include <string.h>
/* include API */
-#include "g711_interface.h"
+#include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h"
/* Runtime statistics */
#include <time.h>
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-24 10:41:42 +0000