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Remove CodecInst pt.2

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The following APIs on AudioCodingModule are deprecated with this CL:
  static int NumberOfCodecs();
  static int Codec(int, CodecInst*);
  static int Codec(const char*, CodecInst*, int, size_t);
  static int Codec(const char*, int, size_t);
  absl::optional<CodecInst> SendCodec() const;
  bool RegisterReceiveCodec(int, const SdpAudioFormat&);
  int RegisterExternalReceiveCodec(int, AudioDecoder*, int, int, const std::string&);
  int UnregisterReceiveCodec(uint8_t);
  int32_t ReceiveCodec(CodecInst*);
  absl::optional<SdpAudioFormat> ReceiveFormat();

As well as this method on RtpRtcp module:
  int32_t RegisterSendPayload(const CodecInst&);

Bug: webrtc:7626
Change-Id: I1230732136f1fe9048cf74afdeab767ca57ac9ce
Reviewed-on: https://webrtc-review.googlesource.com/c/113816
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26025}
This commit is contained in:
Fredrik Solenberg
2018-12-11 12:22:10 +01:00
committed by Commit Bot
parent a134204aa3
commit f693bfae5f
33 changed files with 220 additions and 1625 deletions
Download Patch File Download Diff File Expand all files Collapse all files

68
modules/audio_coding/acm2/audio_coding_module_unittest.cc
View File

@ -19,7 +19,6 @@
#include "api/audio_codecs/opus/audio_encoder_opus.h"
#include "modules/audio_coding/acm2/acm_receive_test.h"
#include "modules/audio_coding/acm2/acm_send_test.h"
#include "modules/audio_coding/codecs/audio_format_conversion.h"
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
#include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
@ -190,12 +189,11 @@ class AudioCodingModuleTestOldApi : public ::testing::Test {
// Set up L16 codec.
virtual void SetUpL16Codec() {
audio_format_ = SdpAudioFormat("L16", kSampleRateHz, 1);
ASSERT_EQ(0, AudioCodingModule::Codec("L16", &codec_, kSampleRateHz, 1));
codec_.pltype = kPayloadType;
pac_size_ = 160;
}
virtual void RegisterCodec() {
EXPECT_EQ(true, acm_->RegisterReceiveCodec(kPayloadType, *audio_format_));
acm_->SetReceiveCodecs({{kPayloadType, *audio_format_}});
acm_->SetEncoder(CreateBuiltinAudioEncoderFactory()->MakeAudioEncoder(
kPayloadType, *audio_format_, absl::nullopt));
}
@ -226,7 +224,7 @@ class AudioCodingModuleTestOldApi : public ::testing::Test {
virtual void VerifyEncoding() {
int last_length = packet_cb_.last_payload_len_bytes();
EXPECT_TRUE(last_length == 2 * codec_.pacsize || last_length == 0)
EXPECT_TRUE(last_length == 2 * pac_size_ || last_length == 0)
<< "Last encoded packet was " << last_length << " bytes.";
}
@ -241,10 +239,8 @@ class AudioCodingModuleTestOldApi : public ::testing::Test {
WebRtcRTPHeader rtp_header_;
AudioFrame input_frame_;
// These two have to be kept in sync for now. In the future, we'll be able to
// eliminate the CodecInst and keep only the SdpAudioFormat.
absl::optional<SdpAudioFormat> audio_format_;
CodecInst codec_;
int pac_size_ = -1;
Clock* clock_;
};
@ -343,7 +339,7 @@ TEST_F(AudioCodingModuleTestOldApi, FailOnZeroDesiredFrequency) {
// Also checks that the frame type is kAudioFrameSpeech.
TEST_F(AudioCodingModuleTestOldApi, TransportCallbackIsInvokedForEachPacket) {
const int k10MsBlocksPerPacket = 3;
codec_.pacsize = k10MsBlocksPerPacket * kSampleRateHz / 100;
pac_size_ = k10MsBlocksPerPacket * kSampleRateHz / 100;
audio_format_->parameters["ptime"] = "30";
RegisterCodec();
const int kLoops = 10;
@ -363,7 +359,7 @@ TEST_F(AudioCodingModuleTestOldApi, TransportCallbackIsInvokedForEachPacket) {
TEST_F(AudioCodingModuleTestOldApi, TimestampSeriesContinuesWhenCodecChanges) {
RegisterCodec(); // This registers the default codec.
uint32_t expected_ts = input_frame_.timestamp_;
int blocks_per_packet = codec_.pacsize / (kSampleRateHz / 100);
int blocks_per_packet = pac_size_ / (kSampleRateHz / 100);
// Encode 5 packets of the first codec type.
const int kNumPackets1 = 5;
for (int j = 0; j < kNumPackets1; ++j) {
@ -373,14 +369,14 @@ TEST_F(AudioCodingModuleTestOldApi, TimestampSeriesContinuesWhenCodecChanges) {
}
EXPECT_EQ(j + 1, packet_cb_.num_calls());
EXPECT_EQ(expected_ts, packet_cb_.last_timestamp());
expected_ts += codec_.pacsize;
expected_ts += pac_size_;
}
// Change codec.
ASSERT_EQ(0, AudioCodingModule::Codec("ISAC", &codec_, kSampleRateHz, 1));
audio_format_ = SdpAudioFormat("ISAC", kSampleRateHz, 1);
pac_size_ = 480;
RegisterCodec();
blocks_per_packet = codec_.pacsize / (kSampleRateHz / 100);
blocks_per_packet = pac_size_ / (kSampleRateHz / 100);
// Encode another 5 packets.
const int kNumPackets2 = 5;
for (int j = 0; j < kNumPackets2; ++j) {
@ -390,7 +386,7 @@ TEST_F(AudioCodingModuleTestOldApi, TimestampSeriesContinuesWhenCodecChanges) {
}
EXPECT_EQ(kNumPackets1 + j + 1, packet_cb_.num_calls());
EXPECT_EQ(expected_ts, packet_cb_.last_timestamp());
expected_ts += codec_.pacsize;
expected_ts += pac_size_;
}
}
#endif
@ -404,9 +400,8 @@ class AudioCodingModuleTestWithComfortNoiseOldApi
: public AudioCodingModuleTestOldApi {
protected:
void RegisterCngCodec(int rtp_payload_type) {
EXPECT_EQ(true,
acm_->RegisterReceiveCodec(
rtp_payload_type, SdpAudioFormat("cn", kSampleRateHz, 1)));
acm_->SetReceiveCodecs({{kPayloadType, *audio_format_},
{rtp_payload_type, {"cn", kSampleRateHz, 1}}});
acm_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* enc) {
AudioEncoderCngConfig config;
config.speech_encoder = std::move(*enc);
@ -461,7 +456,7 @@ class AudioCodingModuleTestWithComfortNoiseOldApi
TEST_F(AudioCodingModuleTestWithComfortNoiseOldApi,
TransportCallbackTestForComfortNoiseRegisterCngLast) {
const int k10MsBlocksPerPacket = 3;
codec_.pacsize = k10MsBlocksPerPacket * kSampleRateHz / 100;
pac_size_ = k10MsBlocksPerPacket * kSampleRateHz / 100;
audio_format_->parameters["ptime"] = "30";
RegisterCodec();
const int kCngPayloadType = 105;
@ -650,12 +645,11 @@ class AcmIsacMtTestOldApi : public AudioCodingModuleMtTestOldApi {
void RegisterCodec() override {
static_assert(kSampleRateHz == 16000, "test designed for iSAC 16 kHz");
audio_format_ = SdpAudioFormat("isac", kSampleRateHz, 1);
AudioCodingModule::Codec("ISAC", &codec_, kSampleRateHz, 1);
codec_.pltype = kPayloadType;
pac_size_ = 480;
// Register iSAC codec in ACM, effectively unregistering the PCM16B codec
// registered in AudioCodingModuleTestOldApi::SetUp();
EXPECT_EQ(true, acm_->RegisterReceiveCodec(kPayloadType, *audio_format_));
acm_->SetReceiveCodecs({{kPayloadType, *audio_format_}});
acm_->SetEncoder(CreateBuiltinAudioEncoderFactory()->MakeAudioEncoder(
kPayloadType, *audio_format_, absl::nullopt));
}
@ -755,15 +749,11 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
}
void RegisterCodec() override {
static_assert(kSampleRateHz == 16000, "test designed for iSAC 16 kHz");
AudioCodingModule::Codec("ISAC", &codec_, kSampleRateHz, 1);
codec_.pltype = kPayloadType;
// Register iSAC codec in ACM, effectively unregistering the PCM16B codec
// registered in AudioCodingModuleTestOldApi::SetUp();
// Only register the decoder for now. The encoder is registered later.
ASSERT_EQ(true, acm_->RegisterReceiveCodec(codec_.pltype,
CodecInstToSdp(codec_)));
static_assert(kSampleRateHz == 16000, "test designed for iSAC 16 kHz");
acm_->SetReceiveCodecs({{kPayloadType, {"ISAC", kSampleRateHz, 1}}});
}
void StartThreads() {
@ -1085,7 +1075,17 @@ TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutputExternalDecoder) {
"bd44bf97e7899186532f91235cef444d",
"9d092dbc96e7ef6870b78c1056e87315"),
factory, [](AudioCodingModule* acm) {
acm->RegisterReceiveCodec(0, {"MockPCMu", 8000, 1});
acm->SetReceiveCodecs({{0, {"MockPCMu", 8000, 1}},
{103, {"ISAC", 16000, 1}},
{104, {"ISAC", 32000, 1}},
{93, {"L16", 8000, 1}},
{94, {"L16", 16000, 1}},
{95, {"L16", 32000, 1}},
{8, {"PCMA", 8000, 1}},
{102, {"ILBC", 8000, 1}},
{13, {"CN", 8000, 1}},
{98, {"CN", 16000, 1}},
{99, {"CN", 32000, 1}}});
});
}
#endif
@ -1746,11 +1746,11 @@ TEST_F(AcmChangeBitRateOldApi, Pcm16_8khz_10ms_32kbps) {
}
TEST_F(AcmSenderBitExactnessOldApi, External_Pcmu_20ms) {
CodecInst codec_inst;
codec_inst.channels = 1;
codec_inst.pacsize = 160;
codec_inst.pltype = 0;
AudioEncoderPcmU encoder(codec_inst);
AudioEncoderPcmU::Config config;
config.frame_size_ms = 20;
config.num_channels = 1;
config.payload_type = 0;
AudioEncoderPcmU encoder(config);
auto mock_encoder = absl::make_unique<MockAudioEncoder>();
// Set expectations on the mock encoder and also delegate the calls to the
// real encoder.
@ -1777,7 +1777,7 @@ TEST_F(AcmSenderBitExactnessOldApi, External_Pcmu_20ms) {
uint32_t, rtc::ArrayView<const int16_t>, rtc::Buffer*)>(
&AudioEncoderPcmU::Encode)));
ASSERT_NO_FATAL_FAILURE(
SetUpTestExternalEncoder(std::move(mock_encoder), codec_inst.pltype));
SetUpTestExternalEncoder(std::move(mock_encoder), config.payload_type));
Run("81a9d4c0bb72e9becc43aef124c981e9", "8f9b8750bd80fe26b6cbf6659b89f0f9",
50, test::AcmReceiveTestOldApi::kMonoOutput);
}