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WebRTC Opus C interface: Add support for non-48 kHz decode sample rate

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Plus tests for 16 kHz.

Bug: webrtc:10631
Change-Id: I2d89bc6d0d9548f0ad7bb1e36d6dfde6b6b31f83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138072
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28099}
This commit is contained in:
Karl Wiberg
2019-05-28 14:41:07 +02:00
committed by Commit Bot
parent 232b6a16cc
commit a1d1a1e976
9 changed files with 170 additions and 135 deletions
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89
modules/audio_coding/codecs/opus/opus_interface.c
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@ -28,15 +28,26 @@ enum {
* side, we must allow for packets of that size. NetEq is currently limited
* to 60 ms on the receive side. */
kWebRtcOpusMaxDecodeFrameSizeMs = 120,
/* Maximum sample count per channel is 48 kHz * maximum frame size in
* milliseconds. */
kWebRtcOpusMaxFrameSizePerChannel = 48 * kWebRtcOpusMaxDecodeFrameSizeMs,
/* Default frame size, 20 ms @ 48 kHz, in samples (for one channel). */
kWebRtcOpusDefaultFrameSize = 960,
};
static int FrameSizePerChannel(int frame_size_ms, int sample_rate_hz) {
RTC_DCHECK_GT(frame_size_ms, 0);
RTC_DCHECK_EQ(frame_size_ms % 10, 0);
RTC_DCHECK_GT(sample_rate_hz, 0);
RTC_DCHECK_EQ(sample_rate_hz % 1000, 0);
return frame_size_ms * (sample_rate_hz / 1000);
}
// Maximum sample count per channel.
static int MaxFrameSizePerChannel(int sample_rate_hz) {
return FrameSizePerChannel(kWebRtcOpusMaxDecodeFrameSizeMs, sample_rate_hz);
}
// Default sample count per channel.
static int DefaultFrameSizePerChannel(int sample_rate_hz) {
return FrameSizePerChannel(20, sample_rate_hz);
}
int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
size_t channels,
int32_t application,
@ -374,7 +385,9 @@ int16_t WebRtcOpus_SetForceChannels(OpusEncInst* inst, size_t num_channels) {
}
}
int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, size_t channels) {
int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst,
size_t channels,
int sample_rate_hz) {
int error;
OpusDecInst* state;
@ -385,14 +398,13 @@ int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, size_t channels) {
return -1;
}
// Create new memory, always at 48000 Hz.
state->decoder = opus_decoder_create(48000,
(int)channels, &error);
state->decoder = opus_decoder_create(sample_rate_hz, (int)channels, &error);
if (error == OPUS_OK && state->decoder) {
// Creation of memory all ok.
state->channels = channels;
state->prev_decoded_samples = kWebRtcOpusDefaultFrameSize;
state->prev_decoded_samples = DefaultFrameSizePerChannel(sample_rate_hz);
state->in_dtx_mode = 0;
state->sample_rate_hz = sample_rate_hz;
*inst = state;
return 0;
}
@ -432,8 +444,9 @@ int16_t WebRtcOpus_MultistreamDecoderCreate(
if (error == OPUS_OK && state->multistream_decoder) {
// Creation of memory all ok.
state->channels = channels;
state->prev_decoded_samples = kWebRtcOpusDefaultFrameSize;
state->prev_decoded_samples = DefaultFrameSizePerChannel(48000);
state->in_dtx_mode = 0;
state->sample_rate_hz = 48000;
*inst = state;
return 0;
}
@ -529,13 +542,9 @@ int WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
*audio_type = DetermineAudioType(inst, encoded_bytes);
decoded_samples = WebRtcOpus_DecodePlc(inst, decoded, 1);
} else {
decoded_samples = DecodeNative(inst,
encoded,
encoded_bytes,
kWebRtcOpusMaxFrameSizePerChannel,
decoded,
audio_type,
0);
decoded_samples = DecodeNative(inst, encoded, encoded_bytes,
MaxFrameSizePerChannel(inst->sample_rate_hz),
decoded, audio_type, 0);
}
if (decoded_samples < 0) {
return -1;
@ -555,10 +564,13 @@ int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
/* The number of samples we ask for is |number_of_lost_frames| times
* |prev_decoded_samples_|. Limit the number of samples to maximum
* |kWebRtcOpusMaxFrameSizePerChannel|. */
* |MaxFrameSizePerChannel()|. */
plc_samples = number_of_lost_frames * inst->prev_decoded_samples;
plc_samples = (plc_samples <= kWebRtcOpusMaxFrameSizePerChannel) ?
plc_samples : kWebRtcOpusMaxFrameSizePerChannel;
const int max_samples_per_channel =
MaxFrameSizePerChannel(inst->sample_rate_hz);
plc_samples = plc_samples <= max_samples_per_channel
? plc_samples
: max_samples_per_channel;
decoded_samples = DecodeNative(inst, NULL, 0, plc_samples,
decoded, &audio_type, 0);
if (decoded_samples < 0) {
@ -578,7 +590,8 @@ int WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
return 0;
}
fec_samples = opus_packet_get_samples_per_frame(encoded, 48000);
fec_samples =
opus_packet_get_samples_per_frame(encoded, inst->sample_rate_hz);
decoded_samples = DecodeNative(inst, encoded, encoded_bytes,
fec_samples, decoded, audio_type, 1);
@ -604,9 +617,10 @@ int WebRtcOpus_DurationEst(OpusDecInst* inst,
/* Invalid payload data. */
return 0;
}
samples = frames * opus_packet_get_samples_per_frame(payload, 48000);
if (samples < 120 || samples > 5760) {
/* Invalid payload duration. */
samples =
frames * opus_packet_get_samples_per_frame(payload, inst->sample_rate_hz);
if (samples > 120 * inst->sample_rate_hz / 1000) {
// More than 120 ms' worth of samples.
return 0;
}
return samples;
@ -615,21 +629,24 @@ int WebRtcOpus_DurationEst(OpusDecInst* inst,
int WebRtcOpus_PlcDuration(OpusDecInst* inst) {
/* The number of samples we ask for is |number_of_lost_frames| times
* |prev_decoded_samples_|. Limit the number of samples to maximum
* |kWebRtcOpusMaxFrameSizePerChannel|. */
* |MaxFrameSizePerChannel()|. */
const int plc_samples = inst->prev_decoded_samples;
return (plc_samples <= kWebRtcOpusMaxFrameSizePerChannel) ?
plc_samples : kWebRtcOpusMaxFrameSizePerChannel;
const int max_samples_per_channel =
MaxFrameSizePerChannel(inst->sample_rate_hz);
return plc_samples <= max_samples_per_channel ? plc_samples
: max_samples_per_channel;
}
int WebRtcOpus_FecDurationEst(const uint8_t* payload,
size_t payload_length_bytes) {
int samples;
size_t payload_length_bytes,
int sample_rate_hz) {
if (WebRtcOpus_PacketHasFec(payload, payload_length_bytes) != 1) {
return 0;
}
samples = opus_packet_get_samples_per_frame(payload, 48000);
if (samples < 480 || samples > 5760) {
const int samples =
opus_packet_get_samples_per_frame(payload, sample_rate_hz);
const int samples_per_ms = sample_rate_hz / 1000;
if (samples < 10 * samples_per_ms || samples > 120 * samples_per_ms) {
/* Invalid payload duration. */
return 0;
}
@ -650,6 +667,8 @@ int WebRtcOpus_PacketHasFec(const uint8_t* payload,
if (payload[0] & 0x80)
return 0;
// For computing the payload length in ms, the sample rate is not important
// since it cancels out. We use 48 kHz, but any valid sample rate would work.
payload_length_ms = opus_packet_get_samples_per_frame(payload, 48000) / 48;
if (10 > payload_length_ms)
payload_length_ms = 10;